henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1 | /* |
kjellander | 65c7f67 | 2016-02-12 00:05:01 -0800 | [diff] [blame] | 2 | * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 3 | * |
kjellander | 65c7f67 | 2016-02-12 00:05:01 -0800 | [diff] [blame] | 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. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 9 | */ |
| 10 | |
kwiberg | 0eb15ed | 2015-12-17 03:04:15 -0800 | [diff] [blame] | 11 | #include <utility> |
| 12 | |
kjellander@webrtc.org | 9b8df25 | 2016-02-12 06:47:59 +0100 | [diff] [blame] | 13 | #include "webrtc/pc/channel.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 14 | |
kjellander | a69d973 | 2016-08-31 07:33:05 -0700 | [diff] [blame] | 15 | #include "webrtc/api/call/audio_sink.h" |
buildbot@webrtc.org | 65b98d1 | 2014-08-07 22:09:08 +0000 | [diff] [blame] | 16 | #include "webrtc/base/bind.h" |
buildbot@webrtc.org | 65b98d1 | 2014-08-07 22:09:08 +0000 | [diff] [blame] | 17 | #include "webrtc/base/byteorder.h" |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 18 | #include "webrtc/base/checks.h" |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 19 | #include "webrtc/base/copyonwritebuffer.h" |
buildbot@webrtc.org | 65b98d1 | 2014-08-07 22:09:08 +0000 | [diff] [blame] | 20 | #include "webrtc/base/dscp.h" |
| 21 | #include "webrtc/base/logging.h" |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 22 | #include "webrtc/base/networkroute.h" |
Peter Boström | 6f28cf0 | 2015-12-07 23:17:15 +0100 | [diff] [blame] | 23 | #include "webrtc/base/trace_event.h" |
kjellander | f475277 | 2016-03-02 05:42:30 -0800 | [diff] [blame] | 24 | #include "webrtc/media/base/mediaconstants.h" |
kjellander | a96e2d7 | 2016-02-04 23:52:28 -0800 | [diff] [blame] | 25 | #include "webrtc/media/base/rtputils.h" |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 26 | #include "webrtc/p2p/base/packettransportinternal.h" |
kjellander@webrtc.org | 9b8df25 | 2016-02-12 06:47:59 +0100 | [diff] [blame] | 27 | #include "webrtc/pc/channelmanager.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 28 | |
| 29 | namespace cricket { |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 30 | using rtc::Bind; |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 31 | |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 32 | namespace { |
kwiberg | 3102294 | 2016-03-11 14:18:21 -0800 | [diff] [blame] | 33 | // See comment below for why we need to use a pointer to a unique_ptr. |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 34 | bool SetRawAudioSink_w(VoiceMediaChannel* channel, |
| 35 | uint32_t ssrc, |
kwiberg | 3102294 | 2016-03-11 14:18:21 -0800 | [diff] [blame] | 36 | std::unique_ptr<webrtc::AudioSinkInterface>* sink) { |
| 37 | channel->SetRawAudioSink(ssrc, std::move(*sink)); |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 38 | return true; |
| 39 | } |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 40 | |
| 41 | struct SendPacketMessageData : public rtc::MessageData { |
| 42 | rtc::CopyOnWriteBuffer packet; |
| 43 | rtc::PacketOptions options; |
| 44 | }; |
| 45 | |
isheriff | 6f8d686 | 2016-05-26 11:24:55 -0700 | [diff] [blame] | 46 | #if defined(ENABLE_EXTERNAL_AUTH) |
| 47 | // Returns the named header extension if found among all extensions, |
| 48 | // nullptr otherwise. |
| 49 | const webrtc::RtpExtension* FindHeaderExtension( |
| 50 | const std::vector<webrtc::RtpExtension>& extensions, |
| 51 | const std::string& uri) { |
| 52 | for (const auto& extension : extensions) { |
| 53 | if (extension.uri == uri) |
| 54 | return &extension; |
| 55 | } |
| 56 | return nullptr; |
| 57 | } |
| 58 | #endif |
| 59 | |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 60 | } // namespace |
| 61 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 62 | enum { |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 63 | MSG_EARLYMEDIATIMEOUT = 1, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 64 | MSG_SEND_RTP_PACKET, |
| 65 | MSG_SEND_RTCP_PACKET, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 66 | MSG_CHANNEL_ERROR, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 67 | MSG_READYTOSENDDATA, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 68 | MSG_DATARECEIVED, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 69 | MSG_FIRSTPACKETRECEIVED, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 70 | }; |
| 71 | |
| 72 | // Value specified in RFC 5764. |
| 73 | static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp"; |
| 74 | |
| 75 | static const int kAgcMinus10db = -10; |
| 76 | |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 77 | static void SafeSetError(const std::string& message, std::string* error_desc) { |
| 78 | if (error_desc) { |
| 79 | *error_desc = message; |
| 80 | } |
| 81 | } |
| 82 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 83 | struct VoiceChannelErrorMessageData : public rtc::MessageData { |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 84 | VoiceChannelErrorMessageData(uint32_t in_ssrc, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 85 | VoiceMediaChannel::Error in_error) |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 86 | : ssrc(in_ssrc), error(in_error) {} |
| 87 | uint32_t ssrc; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 88 | VoiceMediaChannel::Error error; |
| 89 | }; |
| 90 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 91 | struct VideoChannelErrorMessageData : public rtc::MessageData { |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 92 | VideoChannelErrorMessageData(uint32_t in_ssrc, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 93 | VideoMediaChannel::Error in_error) |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 94 | : ssrc(in_ssrc), error(in_error) {} |
| 95 | uint32_t ssrc; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 96 | VideoMediaChannel::Error error; |
| 97 | }; |
| 98 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 99 | struct DataChannelErrorMessageData : public rtc::MessageData { |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 100 | DataChannelErrorMessageData(uint32_t in_ssrc, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 101 | DataMediaChannel::Error in_error) |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 102 | : ssrc(in_ssrc), error(in_error) {} |
| 103 | uint32_t ssrc; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 104 | DataMediaChannel::Error error; |
| 105 | }; |
| 106 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 107 | static const char* PacketType(bool rtcp) { |
| 108 | return (!rtcp) ? "RTP" : "RTCP"; |
| 109 | } |
| 110 | |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 111 | static bool ValidPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 112 | // Check the packet size. We could check the header too if needed. |
| 113 | return (packet && |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 114 | packet->size() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) && |
| 115 | packet->size() <= kMaxRtpPacketLen); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 116 | } |
| 117 | |
| 118 | static bool IsReceiveContentDirection(MediaContentDirection direction) { |
| 119 | return direction == MD_SENDRECV || direction == MD_RECVONLY; |
| 120 | } |
| 121 | |
| 122 | static bool IsSendContentDirection(MediaContentDirection direction) { |
| 123 | return direction == MD_SENDRECV || direction == MD_SENDONLY; |
| 124 | } |
| 125 | |
| 126 | static const MediaContentDescription* GetContentDescription( |
| 127 | const ContentInfo* cinfo) { |
| 128 | if (cinfo == NULL) |
| 129 | return NULL; |
| 130 | return static_cast<const MediaContentDescription*>(cinfo->description); |
| 131 | } |
| 132 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 133 | template <class Codec> |
| 134 | void RtpParametersFromMediaDescription( |
| 135 | const MediaContentDescriptionImpl<Codec>* desc, |
| 136 | RtpParameters<Codec>* params) { |
| 137 | // TODO(pthatcher): Remove this once we're sure no one will give us |
| 138 | // a description without codecs (currently a CA_UPDATE with just |
| 139 | // streams can). |
| 140 | if (desc->has_codecs()) { |
| 141 | params->codecs = desc->codecs(); |
| 142 | } |
| 143 | // TODO(pthatcher): See if we really need |
| 144 | // rtp_header_extensions_set() and remove it if we don't. |
| 145 | if (desc->rtp_header_extensions_set()) { |
| 146 | params->extensions = desc->rtp_header_extensions(); |
| 147 | } |
deadbeef | 1387149 | 2015-12-09 12:37:51 -0800 | [diff] [blame] | 148 | params->rtcp.reduced_size = desc->rtcp_reduced_size(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 149 | } |
| 150 | |
nisse | 0510331 | 2016-03-16 02:22:50 -0700 | [diff] [blame] | 151 | template <class Codec> |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 152 | void RtpSendParametersFromMediaDescription( |
| 153 | const MediaContentDescriptionImpl<Codec>* desc, |
nisse | 0510331 | 2016-03-16 02:22:50 -0700 | [diff] [blame] | 154 | RtpSendParameters<Codec>* send_params) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 155 | RtpParametersFromMediaDescription(desc, send_params); |
| 156 | send_params->max_bandwidth_bps = desc->bandwidth(); |
| 157 | } |
| 158 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 159 | BaseChannel::BaseChannel(rtc::Thread* worker_thread, |
| 160 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 161 | rtc::Thread* signaling_thread, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 162 | MediaChannel* media_channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 163 | const std::string& content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 164 | bool rtcp_mux_required, |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 165 | bool srtp_required) |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 166 | : worker_thread_(worker_thread), |
| 167 | network_thread_(network_thread), |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 168 | signaling_thread_(signaling_thread), |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 169 | content_name_(content_name), |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 170 | rtp_transport_(rtcp_mux_required), |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 171 | srtp_required_(srtp_required), |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 172 | media_channel_(media_channel), |
| 173 | selected_candidate_pair_(nullptr) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 174 | RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
jbauch | dfcab72 | 2017-03-06 00:14:10 -0800 | [diff] [blame] | 175 | #if defined(ENABLE_EXTERNAL_AUTH) |
| 176 | srtp_filter_.EnableExternalAuth(); |
| 177 | #endif |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 178 | LOG(LS_INFO) << "Created channel for " << content_name; |
| 179 | } |
| 180 | |
| 181 | BaseChannel::~BaseChannel() { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 182 | TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 183 | RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 184 | Deinit(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 185 | StopConnectionMonitor(); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 186 | // Eats any outstanding messages or packets. |
| 187 | worker_thread_->Clear(&invoker_); |
| 188 | worker_thread_->Clear(this); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 189 | // We must destroy the media channel before the transport channel, otherwise |
| 190 | // the media channel may try to send on the dead transport channel. NULLing |
| 191 | // is not an effective strategy since the sends will come on another thread. |
| 192 | delete media_channel_; |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 193 | LOG(LS_INFO) << "Destroyed channel: " << content_name_; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 194 | } |
| 195 | |
Danil Chapovalov | dae07ba | 2016-05-14 01:43:50 +0200 | [diff] [blame] | 196 | void BaseChannel::DisconnectTransportChannels_n() { |
| 197 | // Send any outstanding RTCP packets. |
| 198 | FlushRtcpMessages_n(); |
| 199 | |
| 200 | // Stop signals from transport channels, but keep them alive because |
| 201 | // media_channel may use them from a different thread. |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 202 | if (rtp_dtls_transport_) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 203 | DisconnectFromDtlsTransport(rtp_dtls_transport_); |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 204 | } else if (rtp_transport_.rtp_packet_transport()) { |
| 205 | DisconnectFromPacketTransport(rtp_transport_.rtp_packet_transport()); |
Danil Chapovalov | dae07ba | 2016-05-14 01:43:50 +0200 | [diff] [blame] | 206 | } |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 207 | if (rtcp_dtls_transport_) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 208 | DisconnectFromDtlsTransport(rtcp_dtls_transport_); |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 209 | } else if (rtp_transport_.rtcp_packet_transport()) { |
| 210 | DisconnectFromPacketTransport(rtp_transport_.rtcp_packet_transport()); |
Danil Chapovalov | dae07ba | 2016-05-14 01:43:50 +0200 | [diff] [blame] | 211 | } |
| 212 | |
| 213 | // Clear pending read packets/messages. |
| 214 | network_thread_->Clear(&invoker_); |
| 215 | network_thread_->Clear(this); |
| 216 | } |
| 217 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 218 | bool BaseChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport, |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 219 | DtlsTransportInternal* rtcp_dtls_transport, |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 220 | rtc::PacketTransportInternal* rtp_packet_transport, |
| 221 | rtc::PacketTransportInternal* rtcp_packet_transport) { |
skvlad | 6c87a67 | 2016-05-17 17:49:52 -0700 | [diff] [blame] | 222 | if (!network_thread_->Invoke<bool>( |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 223 | RTC_FROM_HERE, Bind(&BaseChannel::InitNetwork_n, this, |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 224 | rtp_dtls_transport, rtcp_dtls_transport, |
| 225 | rtp_packet_transport, rtcp_packet_transport))) { |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 226 | return false; |
| 227 | } |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 228 | // Both RTP and RTCP channels should be set, we can call SetInterface on |
| 229 | // the media channel and it can set network options. |
| 230 | RTC_DCHECK_RUN_ON(worker_thread_); |
wu@webrtc.org | de30501 | 2013-10-31 15:40:38 +0000 | [diff] [blame] | 231 | media_channel_->SetInterface(this); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 232 | return true; |
| 233 | } |
| 234 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 235 | bool BaseChannel::InitNetwork_n( |
| 236 | DtlsTransportInternal* rtp_dtls_transport, |
| 237 | DtlsTransportInternal* rtcp_dtls_transport, |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 238 | rtc::PacketTransportInternal* rtp_packet_transport, |
| 239 | rtc::PacketTransportInternal* rtcp_packet_transport) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 240 | RTC_DCHECK(network_thread_->IsCurrent()); |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 241 | SetTransports_n(rtp_dtls_transport, rtcp_dtls_transport, rtp_packet_transport, |
| 242 | rtcp_packet_transport); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 243 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 244 | if (rtp_dtls_transport_ && |
| 245 | !SetDtlsSrtpCryptoSuites_n(rtp_dtls_transport_, false)) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 246 | return false; |
| 247 | } |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 248 | if (rtcp_dtls_transport_ && |
| 249 | !SetDtlsSrtpCryptoSuites_n(rtcp_dtls_transport_, true)) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 250 | return false; |
| 251 | } |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 252 | if (rtp_transport_.rtcp_mux_required()) { |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 253 | rtcp_mux_filter_.SetActive(); |
| 254 | } |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 255 | return true; |
| 256 | } |
| 257 | |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 258 | void BaseChannel::Deinit() { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 259 | RTC_DCHECK(worker_thread_->IsCurrent()); |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 260 | media_channel_->SetInterface(NULL); |
Danil Chapovalov | dae07ba | 2016-05-14 01:43:50 +0200 | [diff] [blame] | 261 | // Packets arrive on the network thread, processing packets calls virtual |
| 262 | // functions, so need to stop this process in Deinit that is called in |
| 263 | // derived classes destructor. |
| 264 | network_thread_->Invoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 265 | RTC_FROM_HERE, Bind(&BaseChannel::DisconnectTransportChannels_n, this)); |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 266 | } |
| 267 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 268 | void BaseChannel::SetTransports(DtlsTransportInternal* rtp_dtls_transport, |
| 269 | DtlsTransportInternal* rtcp_dtls_transport) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 270 | network_thread_->Invoke<void>( |
| 271 | RTC_FROM_HERE, |
| 272 | Bind(&BaseChannel::SetTransports_n, this, rtp_dtls_transport, |
| 273 | rtcp_dtls_transport, rtp_dtls_transport, rtcp_dtls_transport)); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 274 | } |
| 275 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 276 | void BaseChannel::SetTransports( |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 277 | rtc::PacketTransportInternal* rtp_packet_transport, |
| 278 | rtc::PacketTransportInternal* rtcp_packet_transport) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 279 | network_thread_->Invoke<void>( |
| 280 | RTC_FROM_HERE, Bind(&BaseChannel::SetTransports_n, this, nullptr, nullptr, |
| 281 | rtp_packet_transport, rtcp_packet_transport)); |
| 282 | } |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 283 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 284 | void BaseChannel::SetTransports_n( |
| 285 | DtlsTransportInternal* rtp_dtls_transport, |
| 286 | DtlsTransportInternal* rtcp_dtls_transport, |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 287 | rtc::PacketTransportInternal* rtp_packet_transport, |
| 288 | rtc::PacketTransportInternal* rtcp_packet_transport) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 289 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 290 | // Validate some assertions about the input. |
| 291 | RTC_DCHECK(rtp_packet_transport); |
| 292 | RTC_DCHECK_EQ(NeedsRtcpTransport(), rtcp_packet_transport != nullptr); |
| 293 | if (rtp_dtls_transport || rtcp_dtls_transport) { |
| 294 | // DTLS/non-DTLS pointers should be to the same object. |
| 295 | RTC_DCHECK(rtp_dtls_transport == rtp_packet_transport); |
| 296 | RTC_DCHECK(rtcp_dtls_transport == rtcp_packet_transport); |
| 297 | // Can't go from non-DTLS to DTLS. |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 298 | RTC_DCHECK(!rtp_transport_.rtp_packet_transport() || rtp_dtls_transport_); |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 299 | } else { |
| 300 | // Can't go from DTLS to non-DTLS. |
| 301 | RTC_DCHECK(!rtp_dtls_transport_); |
| 302 | } |
| 303 | // Transport names should be the same. |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 304 | if (rtp_dtls_transport && rtcp_dtls_transport) { |
| 305 | RTC_DCHECK(rtp_dtls_transport->transport_name() == |
| 306 | rtcp_dtls_transport->transport_name()); |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 307 | } |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 308 | std::string debug_name; |
| 309 | if (rtp_dtls_transport) { |
| 310 | transport_name_ = rtp_dtls_transport->transport_name(); |
| 311 | debug_name = transport_name_; |
| 312 | } else { |
| 313 | debug_name = rtp_packet_transport->debug_name(); |
| 314 | } |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 315 | if (rtp_packet_transport == rtp_transport_.rtp_packet_transport()) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 316 | // Nothing to do if transport isn't changing. |
deadbeef | bad5dad | 2017-01-17 18:32:35 -0800 | [diff] [blame] | 317 | return; |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 318 | } |
| 319 | |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 320 | // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport |
| 321 | // changes and wait until the DTLS handshake is complete to set the newly |
| 322 | // negotiated parameters. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 323 | if (ShouldSetupDtlsSrtp_n()) { |
guoweis | 4638331 | 2015-12-17 16:45:59 -0800 | [diff] [blame] | 324 | // Set |writable_| to false such that UpdateWritableState_w can set up |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 325 | // DTLS-SRTP when |writable_| becomes true again. |
guoweis | 4638331 | 2015-12-17 16:45:59 -0800 | [diff] [blame] | 326 | writable_ = false; |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 327 | srtp_filter_.ResetParams(); |
| 328 | } |
| 329 | |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 330 | // If this BaseChannel doesn't require RTCP mux and we haven't fully |
| 331 | // negotiated RTCP mux, we need an RTCP transport. |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 332 | if (rtcp_packet_transport) { |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 333 | LOG(LS_INFO) << "Setting RTCP Transport for " << content_name() << " on " |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 334 | << debug_name << " transport " << rtcp_packet_transport; |
| 335 | SetTransport_n(true, rtcp_dtls_transport, rtcp_packet_transport); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 336 | } |
| 337 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 338 | LOG(LS_INFO) << "Setting RTP Transport for " << content_name() << " on " |
| 339 | << debug_name << " transport " << rtp_packet_transport; |
| 340 | SetTransport_n(false, rtp_dtls_transport, rtp_packet_transport); |
guoweis | 4638331 | 2015-12-17 16:45:59 -0800 | [diff] [blame] | 341 | |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 342 | // Update aggregate writable/ready-to-send state between RTP and RTCP upon |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 343 | // setting new transport channels. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 344 | UpdateWritableState_n(); |
deadbeef | 062ce9f | 2016-08-26 21:42:15 -0700 | [diff] [blame] | 345 | // We can only update ready-to-send after updating writability. |
| 346 | // |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 347 | // On setting a new channel, assume it's ready to send if it's writable, |
| 348 | // because we have no way of knowing otherwise (the channel doesn't give us |
| 349 | // "was last send successful?"). |
| 350 | // |
| 351 | // This won't always be accurate (the last SendPacket call from another |
| 352 | // BaseChannel could have resulted in an error), but even so, we'll just |
| 353 | // encounter the error again and update "ready to send" accordingly. |
deadbeef | 062ce9f | 2016-08-26 21:42:15 -0700 | [diff] [blame] | 354 | SetTransportChannelReadyToSend( |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 355 | false, rtp_packet_transport && rtp_packet_transport->writable()); |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 356 | SetTransportChannelReadyToSend( |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 357 | true, rtcp_packet_transport && rtcp_packet_transport->writable()); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 358 | } |
| 359 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 360 | void BaseChannel::SetTransport_n( |
| 361 | bool rtcp, |
| 362 | DtlsTransportInternal* new_dtls_transport, |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 363 | rtc::PacketTransportInternal* new_packet_transport) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 364 | RTC_DCHECK(network_thread_->IsCurrent()); |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 365 | DtlsTransportInternal*& old_dtls_transport = |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 366 | rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_; |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 367 | rtc::PacketTransportInternal* old_packet_transport = |
| 368 | rtcp ? rtp_transport_.rtcp_packet_transport() |
| 369 | : rtp_transport_.rtp_packet_transport(); |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 370 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 371 | if (!old_packet_transport && !new_packet_transport) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 372 | // Nothing to do. |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 373 | return; |
| 374 | } |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 375 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 376 | RTC_DCHECK(old_packet_transport != new_packet_transport); |
| 377 | if (old_dtls_transport) { |
| 378 | DisconnectFromDtlsTransport(old_dtls_transport); |
| 379 | } else if (old_packet_transport) { |
| 380 | DisconnectFromPacketTransport(old_packet_transport); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 381 | } |
| 382 | |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 383 | if (rtcp) { |
| 384 | rtp_transport_.set_rtcp_packet_transport(new_packet_transport); |
| 385 | } else { |
| 386 | rtp_transport_.set_rtp_packet_transport(new_packet_transport); |
| 387 | } |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 388 | old_dtls_transport = new_dtls_transport; |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 389 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 390 | // If there's no new transport, we're done after disconnecting from old one. |
| 391 | if (!new_packet_transport) { |
| 392 | return; |
| 393 | } |
| 394 | |
| 395 | if (rtcp && new_dtls_transport) { |
| 396 | RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_filter_.IsActive())) |
| 397 | << "Setting RTCP for DTLS/SRTP after SrtpFilter is active " |
| 398 | << "should never happen."; |
| 399 | } |
| 400 | if (new_dtls_transport) { |
| 401 | ConnectToDtlsTransport(new_dtls_transport); |
| 402 | } else { |
| 403 | ConnectToPacketTransport(new_packet_transport); |
| 404 | } |
| 405 | auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_; |
| 406 | for (const auto& pair : socket_options) { |
| 407 | new_packet_transport->SetOption(pair.first, pair.second); |
guoweis | 4638331 | 2015-12-17 16:45:59 -0800 | [diff] [blame] | 408 | } |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 409 | } |
| 410 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 411 | void BaseChannel::ConnectToDtlsTransport(DtlsTransportInternal* transport) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 412 | RTC_DCHECK(network_thread_->IsCurrent()); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 413 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 414 | transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState); |
| 415 | transport->SignalReadPacket.connect(this, &BaseChannel::OnPacketRead); |
| 416 | transport->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend); |
| 417 | transport->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState); |
| 418 | transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); |
| 419 | transport->ice_transport()->SignalSelectedCandidatePairChanged.connect( |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 420 | this, &BaseChannel::OnSelectedCandidatePairChanged); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 421 | } |
| 422 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 423 | void BaseChannel::DisconnectFromDtlsTransport( |
| 424 | DtlsTransportInternal* transport) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 425 | RTC_DCHECK(network_thread_->IsCurrent()); |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 426 | OnSelectedCandidatePairChanged(transport->ice_transport(), nullptr, -1, |
| 427 | false); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 428 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 429 | transport->SignalWritableState.disconnect(this); |
| 430 | transport->SignalReadPacket.disconnect(this); |
| 431 | transport->SignalReadyToSend.disconnect(this); |
| 432 | transport->SignalDtlsState.disconnect(this); |
| 433 | transport->SignalSentPacket.disconnect(this); |
| 434 | transport->ice_transport()->SignalSelectedCandidatePairChanged.disconnect( |
| 435 | this); |
pthatcher@webrtc.org | 6ad507a | 2015-03-16 20:19:12 +0000 | [diff] [blame] | 436 | } |
| 437 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 438 | void BaseChannel::ConnectToPacketTransport( |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 439 | rtc::PacketTransportInternal* transport) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 440 | RTC_DCHECK_RUN_ON(network_thread_); |
| 441 | transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState); |
| 442 | transport->SignalReadPacket.connect(this, &BaseChannel::OnPacketRead); |
| 443 | transport->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend); |
| 444 | transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); |
| 445 | } |
| 446 | |
| 447 | void BaseChannel::DisconnectFromPacketTransport( |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 448 | rtc::PacketTransportInternal* transport) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 449 | RTC_DCHECK_RUN_ON(network_thread_); |
| 450 | transport->SignalWritableState.disconnect(this); |
| 451 | transport->SignalReadPacket.disconnect(this); |
| 452 | transport->SignalReadyToSend.disconnect(this); |
| 453 | transport->SignalSentPacket.disconnect(this); |
| 454 | } |
| 455 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 456 | bool BaseChannel::Enable(bool enable) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 457 | worker_thread_->Invoke<void>( |
| 458 | RTC_FROM_HERE, |
| 459 | Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w, |
| 460 | this)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 461 | return true; |
| 462 | } |
| 463 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 464 | bool BaseChannel::AddRecvStream(const StreamParams& sp) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 465 | return InvokeOnWorker(RTC_FROM_HERE, |
| 466 | Bind(&BaseChannel::AddRecvStream_w, this, sp)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 467 | } |
| 468 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 469 | bool BaseChannel::RemoveRecvStream(uint32_t ssrc) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 470 | return InvokeOnWorker(RTC_FROM_HERE, |
| 471 | Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 472 | } |
| 473 | |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 474 | bool BaseChannel::AddSendStream(const StreamParams& sp) { |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 475 | return InvokeOnWorker( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 476 | RTC_FROM_HERE, Bind(&MediaChannel::AddSendStream, media_channel(), sp)); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 477 | } |
| 478 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 479 | bool BaseChannel::RemoveSendStream(uint32_t ssrc) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 480 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&MediaChannel::RemoveSendStream, |
| 481 | media_channel(), ssrc)); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 482 | } |
| 483 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 484 | bool BaseChannel::SetLocalContent(const MediaContentDescription* content, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 485 | ContentAction action, |
| 486 | std::string* error_desc) { |
Peter Boström | 9f45a45 | 2015-12-08 13:25:57 +0100 | [diff] [blame] | 487 | TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent"); |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 488 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&BaseChannel::SetLocalContent_w, |
| 489 | this, content, action, error_desc)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 490 | } |
| 491 | |
| 492 | bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 493 | ContentAction action, |
| 494 | std::string* error_desc) { |
Peter Boström | 9f45a45 | 2015-12-08 13:25:57 +0100 | [diff] [blame] | 495 | TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent"); |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 496 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w, |
| 497 | this, content, action, error_desc)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 498 | } |
| 499 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 500 | void BaseChannel::StartConnectionMonitor(int cms) { |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 501 | // We pass in the BaseChannel instead of the rtp_dtls_transport_ |
| 502 | // because if the rtp_dtls_transport_ changes, the ConnectionMonitor |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 503 | // would be pointing to the wrong TransportChannel. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 504 | // We pass in the network thread because on that thread connection monitor |
| 505 | // will call BaseChannel::GetConnectionStats which must be called on the |
| 506 | // network thread. |
| 507 | connection_monitor_.reset( |
| 508 | new ConnectionMonitor(this, network_thread(), rtc::Thread::Current())); |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 509 | connection_monitor_->SignalUpdate.connect( |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 510 | this, &BaseChannel::OnConnectionMonitorUpdate); |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 511 | connection_monitor_->Start(cms); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 512 | } |
| 513 | |
| 514 | void BaseChannel::StopConnectionMonitor() { |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 515 | if (connection_monitor_) { |
| 516 | connection_monitor_->Stop(); |
| 517 | connection_monitor_.reset(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 518 | } |
| 519 | } |
| 520 | |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 521 | bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 522 | RTC_DCHECK(network_thread_->IsCurrent()); |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 523 | if (!rtp_dtls_transport_) { |
| 524 | return false; |
| 525 | } |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 526 | return rtp_dtls_transport_->ice_transport()->GetStats(infos); |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 527 | } |
| 528 | |
| 529 | bool BaseChannel::NeedsRtcpTransport() { |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 530 | // If this BaseChannel doesn't require RTCP mux and we haven't fully |
| 531 | // negotiated RTCP mux, we need an RTCP transport. |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 532 | return !rtp_transport_.rtcp_mux_required() && |
| 533 | !rtcp_mux_filter_.IsFullyActive(); |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 534 | } |
| 535 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 536 | bool BaseChannel::IsReadyToReceiveMedia_w() const { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 537 | // Receive data if we are enabled and have local content, |
| 538 | return enabled() && IsReceiveContentDirection(local_content_direction_); |
| 539 | } |
| 540 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 541 | bool BaseChannel::IsReadyToSendMedia_w() const { |
| 542 | // Need to access some state updated on the network thread. |
| 543 | return network_thread_->Invoke<bool>( |
| 544 | RTC_FROM_HERE, Bind(&BaseChannel::IsReadyToSendMedia_n, this)); |
| 545 | } |
| 546 | |
| 547 | bool BaseChannel::IsReadyToSendMedia_n() const { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 548 | // Send outgoing data if we are enabled, have local and remote content, |
| 549 | // and we have had some form of connectivity. |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 550 | return enabled() && IsReceiveContentDirection(remote_content_direction_) && |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 551 | IsSendContentDirection(local_content_direction_) && |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 552 | was_ever_writable() && |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 553 | (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp_n()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 554 | } |
| 555 | |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 556 | bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet, |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 557 | const rtc::PacketOptions& options) { |
| 558 | return SendPacket(false, packet, options); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 559 | } |
| 560 | |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 561 | bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet, |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 562 | const rtc::PacketOptions& options) { |
| 563 | return SendPacket(true, packet, options); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 564 | } |
| 565 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 566 | int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 567 | int value) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 568 | return network_thread_->Invoke<int>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 569 | RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value)); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 570 | } |
| 571 | |
| 572 | int BaseChannel::SetOption_n(SocketType type, |
| 573 | rtc::Socket::Option opt, |
| 574 | int value) { |
| 575 | RTC_DCHECK(network_thread_->IsCurrent()); |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 576 | rtc::PacketTransportInternal* transport = nullptr; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 577 | switch (type) { |
wu@webrtc.org | 9caf276 | 2013-12-11 18:25:07 +0000 | [diff] [blame] | 578 | case ST_RTP: |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 579 | transport = rtp_transport_.rtp_packet_transport(); |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 580 | socket_options_.push_back( |
| 581 | std::pair<rtc::Socket::Option, int>(opt, value)); |
wu@webrtc.org | 9caf276 | 2013-12-11 18:25:07 +0000 | [diff] [blame] | 582 | break; |
| 583 | case ST_RTCP: |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 584 | transport = rtp_transport_.rtcp_packet_transport(); |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 585 | rtcp_socket_options_.push_back( |
| 586 | std::pair<rtc::Socket::Option, int>(opt, value)); |
wu@webrtc.org | 9caf276 | 2013-12-11 18:25:07 +0000 | [diff] [blame] | 587 | break; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 588 | } |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 589 | return transport ? transport->SetOption(opt, value) : -1; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 590 | } |
| 591 | |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 592 | bool BaseChannel::SetCryptoOptions(const rtc::CryptoOptions& crypto_options) { |
| 593 | crypto_options_ = crypto_options; |
| 594 | return true; |
| 595 | } |
| 596 | |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 597 | void BaseChannel::OnWritableState(rtc::PacketTransportInternal* transport) { |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 598 | RTC_DCHECK(transport == rtp_transport_.rtp_packet_transport() || |
| 599 | transport == rtp_transport_.rtcp_packet_transport()); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 600 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 601 | UpdateWritableState_n(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 602 | } |
| 603 | |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 604 | void BaseChannel::OnPacketRead(rtc::PacketTransportInternal* transport, |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 605 | const char* data, |
| 606 | size_t len, |
| 607 | const rtc::PacketTime& packet_time, |
| 608 | int flags) { |
| 609 | TRACE_EVENT0("webrtc", "BaseChannel::OnPacketRead"); |
| 610 | // OnPacketRead gets called from P2PSocket; now pass data to MediaEngine |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 611 | RTC_DCHECK(network_thread_->IsCurrent()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 612 | |
| 613 | // When using RTCP multiplexing we might get RTCP packets on the RTP |
| 614 | // transport. We feed RTP traffic into the demuxer to determine if it is RTCP. |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 615 | bool rtcp = PacketIsRtcp(transport, data, len); |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 616 | rtc::CopyOnWriteBuffer packet(data, len); |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 +0000 | [diff] [blame] | 617 | HandlePacket(rtcp, &packet, packet_time); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 618 | } |
| 619 | |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 620 | void BaseChannel::OnReadyToSend(rtc::PacketTransportInternal* transport) { |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 621 | RTC_DCHECK(transport == rtp_transport_.rtp_packet_transport() || |
| 622 | transport == rtp_transport_.rtcp_packet_transport()); |
| 623 | SetTransportChannelReadyToSend( |
| 624 | transport == rtp_transport_.rtcp_packet_transport(), true); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 625 | } |
| 626 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 627 | void BaseChannel::OnDtlsState(DtlsTransportInternal* transport, |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 628 | DtlsTransportState state) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 629 | if (!ShouldSetupDtlsSrtp_n()) { |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 630 | return; |
| 631 | } |
| 632 | |
| 633 | // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED |
| 634 | // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 635 | // cover other scenarios like the whole transport is writable (not just this |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 636 | // TransportChannel) or when TransportChannel is attached after DTLS is |
| 637 | // negotiated. |
| 638 | if (state != DTLS_TRANSPORT_CONNECTED) { |
| 639 | srtp_filter_.ResetParams(); |
| 640 | } |
| 641 | } |
| 642 | |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 643 | void BaseChannel::OnSelectedCandidatePairChanged( |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 644 | IceTransportInternal* ice_transport, |
Honghai Zhang | 52dce73 | 2016-03-31 12:37:31 -0700 | [diff] [blame] | 645 | CandidatePairInterface* selected_candidate_pair, |
Taylor Brandstetter | 6bb1ef2 | 2016-06-27 18:09:03 -0700 | [diff] [blame] | 646 | int last_sent_packet_id, |
| 647 | bool ready_to_send) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 648 | RTC_DCHECK((rtp_dtls_transport_ && |
| 649 | ice_transport == rtp_dtls_transport_->ice_transport()) || |
| 650 | (rtcp_dtls_transport_ && |
| 651 | ice_transport == rtcp_dtls_transport_->ice_transport())); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 652 | RTC_DCHECK(network_thread_->IsCurrent()); |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 653 | selected_candidate_pair_ = selected_candidate_pair; |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 654 | std::string transport_name = ice_transport->transport_name(); |
Honghai Zhang | 0e533ef | 2016-04-19 15:41:36 -0700 | [diff] [blame] | 655 | rtc::NetworkRoute network_route; |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 656 | if (selected_candidate_pair) { |
Honghai Zhang | 0e533ef | 2016-04-19 15:41:36 -0700 | [diff] [blame] | 657 | network_route = rtc::NetworkRoute( |
Taylor Brandstetter | 6bb1ef2 | 2016-06-27 18:09:03 -0700 | [diff] [blame] | 658 | ready_to_send, selected_candidate_pair->local_candidate().network_id(), |
Honghai Zhang | 0e533ef | 2016-04-19 15:41:36 -0700 | [diff] [blame] | 659 | selected_candidate_pair->remote_candidate().network_id(), |
| 660 | last_sent_packet_id); |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 661 | |
| 662 | UpdateTransportOverhead(); |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 663 | } |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 664 | invoker_.AsyncInvoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 665 | RTC_FROM_HERE, worker_thread_, |
| 666 | Bind(&MediaChannel::OnNetworkRouteChanged, media_channel_, transport_name, |
| 667 | network_route)); |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 668 | } |
| 669 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 670 | void BaseChannel::SetTransportChannelReadyToSend(bool rtcp, bool ready) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 671 | RTC_DCHECK(network_thread_->IsCurrent()); |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 672 | if (rtcp) { |
| 673 | rtcp_ready_to_send_ = ready; |
| 674 | } else { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 675 | rtp_ready_to_send_ = ready; |
| 676 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 677 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 678 | bool ready_to_send = |
| 679 | (rtp_ready_to_send_ && |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 680 | // In the case of rtcp mux |rtcp_packet_transport_| will be null. |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 681 | (rtcp_ready_to_send_ || !rtp_transport_.rtcp_packet_transport())); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 682 | |
| 683 | invoker_.AsyncInvoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 684 | RTC_FROM_HERE, worker_thread_, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 685 | Bind(&MediaChannel::OnReadyToSend, media_channel_, ready_to_send)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 686 | } |
| 687 | |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 688 | bool BaseChannel::PacketIsRtcp(const rtc::PacketTransportInternal* transport, |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 689 | const char* data, |
| 690 | size_t len) { |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 691 | return (transport == rtp_transport_.rtcp_packet_transport() || |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 +0000 | [diff] [blame] | 692 | rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len))); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 693 | } |
| 694 | |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 695 | bool BaseChannel::SendPacket(bool rtcp, |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 696 | rtc::CopyOnWriteBuffer* packet, |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 697 | const rtc::PacketOptions& options) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 698 | // SendPacket gets called from MediaEngine, on a pacer or an encoder thread. |
| 699 | // If the thread is not our network thread, we will post to our network |
| 700 | // so that the real work happens on our network. This avoids us having to |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 701 | // synchronize access to all the pieces of the send path, including |
| 702 | // SRTP and the inner workings of the transport channels. |
| 703 | // The only downside is that we can't return a proper failure code if |
| 704 | // needed. Since UDP is unreliable anyway, this should be a non-issue. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 705 | if (!network_thread_->IsCurrent()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 706 | // Avoid a copy by transferring the ownership of the packet data. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 707 | int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET; |
| 708 | SendPacketMessageData* data = new SendPacketMessageData; |
kwiberg | 0eb15ed | 2015-12-17 03:04:15 -0800 | [diff] [blame] | 709 | data->packet = std::move(*packet); |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 710 | data->options = options; |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 711 | network_thread_->Post(RTC_FROM_HERE, this, message_id, data); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 712 | return true; |
| 713 | } |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 714 | TRACE_EVENT0("webrtc", "BaseChannel::SendPacket"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 715 | |
| 716 | // Now that we are on the correct thread, ensure we have a place to send this |
| 717 | // packet before doing anything. (We might get RTCP packets that we don't |
| 718 | // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP |
| 719 | // transport. |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 720 | rtc::PacketTransportInternal* transport = |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 721 | (!rtcp || rtcp_mux_filter_.IsActive()) |
| 722 | ? rtp_transport_.rtp_packet_transport() |
| 723 | : rtp_transport_.rtcp_packet_transport(); |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 724 | if (!transport || !transport->writable()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 725 | return false; |
| 726 | } |
| 727 | |
| 728 | // Protect ourselves against crazy data. |
| 729 | if (!ValidPacket(rtcp, packet)) { |
| 730 | LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " " |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 731 | << PacketType(rtcp) |
| 732 | << " packet: wrong size=" << packet->size(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 733 | return false; |
| 734 | } |
| 735 | |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 736 | rtc::PacketOptions updated_options; |
| 737 | updated_options = options; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 738 | // Protect if needed. |
| 739 | if (srtp_filter_.IsActive()) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 740 | TRACE_EVENT0("webrtc", "SRTP Encode"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 741 | bool res; |
Karl Wiberg | c56ac1e | 2015-05-04 14:54:55 +0200 | [diff] [blame] | 742 | uint8_t* data = packet->data(); |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 743 | int len = static_cast<int>(packet->size()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 744 | if (!rtcp) { |
henrike@webrtc.org | d43aa9d | 2014-02-21 23:43:24 +0000 | [diff] [blame] | 745 | // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done |
| 746 | // inside libsrtp for a RTP packet. A external HMAC module will be writing |
| 747 | // a fake HMAC value. This is ONLY done for a RTP packet. |
| 748 | // Socket layer will update rtp sendtime extension header if present in |
| 749 | // packet with current time before updating the HMAC. |
| 750 | #if !defined(ENABLE_EXTERNAL_AUTH) |
| 751 | res = srtp_filter_.ProtectRtp( |
| 752 | data, len, static_cast<int>(packet->capacity()), &len); |
| 753 | #else |
jbauch | d48f488 | 2017-03-01 15:34:36 -0800 | [diff] [blame] | 754 | if (!srtp_filter_.IsExternalAuthActive()) { |
| 755 | res = srtp_filter_.ProtectRtp( |
| 756 | data, len, static_cast<int>(packet->capacity()), &len); |
| 757 | } else { |
| 758 | updated_options.packet_time_params.rtp_sendtime_extension_id = |
| 759 | rtp_abs_sendtime_extn_id_; |
| 760 | res = srtp_filter_.ProtectRtp( |
| 761 | data, len, static_cast<int>(packet->capacity()), &len, |
| 762 | &updated_options.packet_time_params.srtp_packet_index); |
| 763 | // If protection succeeds, let's get auth params from srtp. |
henrike@webrtc.org | d43aa9d | 2014-02-21 23:43:24 +0000 | [diff] [blame] | 764 | if (res) { |
jbauch | d48f488 | 2017-03-01 15:34:36 -0800 | [diff] [blame] | 765 | uint8_t* auth_key = NULL; |
| 766 | int key_len; |
| 767 | res = srtp_filter_.GetRtpAuthParams( |
| 768 | &auth_key, &key_len, |
| 769 | &updated_options.packet_time_params.srtp_auth_tag_len); |
| 770 | if (res) { |
| 771 | updated_options.packet_time_params.srtp_auth_key.resize(key_len); |
| 772 | updated_options.packet_time_params.srtp_auth_key.assign( |
| 773 | auth_key, auth_key + key_len); |
| 774 | } |
henrike@webrtc.org | d43aa9d | 2014-02-21 23:43:24 +0000 | [diff] [blame] | 775 | } |
| 776 | } |
| 777 | #endif |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 778 | if (!res) { |
| 779 | int seq_num = -1; |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 780 | uint32_t ssrc = 0; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 781 | GetRtpSeqNum(data, len, &seq_num); |
| 782 | GetRtpSsrc(data, len, &ssrc); |
| 783 | LOG(LS_ERROR) << "Failed to protect " << content_name_ |
| 784 | << " RTP packet: size=" << len |
| 785 | << ", seqnum=" << seq_num << ", SSRC=" << ssrc; |
| 786 | return false; |
| 787 | } |
| 788 | } else { |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 +0000 | [diff] [blame] | 789 | res = srtp_filter_.ProtectRtcp(data, len, |
| 790 | static_cast<int>(packet->capacity()), |
| 791 | &len); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 792 | if (!res) { |
| 793 | int type = -1; |
| 794 | GetRtcpType(data, len, &type); |
| 795 | LOG(LS_ERROR) << "Failed to protect " << content_name_ |
| 796 | << " RTCP packet: size=" << len << ", type=" << type; |
| 797 | return false; |
| 798 | } |
| 799 | } |
| 800 | |
| 801 | // Update the length of the packet now that we've added the auth tag. |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 802 | packet->SetSize(len); |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 803 | } else if (srtp_required_) { |
deadbeef | 8f425f9 | 2016-12-01 12:26:27 -0800 | [diff] [blame] | 804 | // The audio/video engines may attempt to send RTCP packets as soon as the |
| 805 | // streams are created, so don't treat this as an error for RTCP. |
| 806 | // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809 |
| 807 | if (rtcp) { |
| 808 | return false; |
| 809 | } |
| 810 | // However, there shouldn't be any RTP packets sent before SRTP is set up |
| 811 | // (and SetSend(true) is called). |
| 812 | LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive" |
| 813 | << " and crypto is required"; |
nisse | eb4ca4e | 2017-01-12 02:24:27 -0800 | [diff] [blame] | 814 | RTC_NOTREACHED(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 815 | return false; |
| 816 | } |
| 817 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 818 | // Bon voyage. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 819 | int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL; |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 820 | int ret = transport->SendPacket(packet->data<char>(), packet->size(), |
| 821 | updated_options, flags); |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 822 | if (ret != static_cast<int>(packet->size())) { |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 823 | if (transport->GetError() == ENOTCONN) { |
skvlad | c309e0e | 2016-07-28 17:15:20 -0700 | [diff] [blame] | 824 | LOG(LS_WARNING) << "Got ENOTCONN from transport."; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 825 | SetTransportChannelReadyToSend(rtcp, false); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 826 | } |
| 827 | return false; |
| 828 | } |
| 829 | return true; |
| 830 | } |
| 831 | |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 832 | bool BaseChannel::WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 833 | // Protect ourselves against crazy data. |
| 834 | if (!ValidPacket(rtcp, packet)) { |
| 835 | LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " " |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 836 | << PacketType(rtcp) |
| 837 | << " packet: wrong size=" << packet->size(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 838 | return false; |
| 839 | } |
pbos | 482b12e | 2015-11-16 10:19:58 -0800 | [diff] [blame] | 840 | if (rtcp) { |
| 841 | // Permit all (seemingly valid) RTCP packets. |
| 842 | return true; |
| 843 | } |
| 844 | // Check whether we handle this payload. |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 845 | return bundle_filter_.DemuxPacket(packet->data(), packet->size()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 846 | } |
| 847 | |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 848 | void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 849 | const rtc::PacketTime& packet_time) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 850 | RTC_DCHECK(network_thread_->IsCurrent()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 851 | if (!WantsPacket(rtcp, packet)) { |
| 852 | return; |
| 853 | } |
| 854 | |
honghaiz@google.com | a67ca1a | 2015-01-28 19:48:33 +0000 | [diff] [blame] | 855 | // We are only interested in the first rtp packet because that |
| 856 | // indicates the media has started flowing. |
| 857 | if (!has_received_packet_ && !rtcp) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 858 | has_received_packet_ = true; |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 859 | signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 860 | } |
| 861 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 862 | // Unprotect the packet, if needed. |
| 863 | if (srtp_filter_.IsActive()) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 864 | TRACE_EVENT0("webrtc", "SRTP Decode"); |
Karl Wiberg | 9478437 | 2015-04-20 14:03:07 +0200 | [diff] [blame] | 865 | char* data = packet->data<char>(); |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 866 | int len = static_cast<int>(packet->size()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 867 | bool res; |
| 868 | if (!rtcp) { |
| 869 | res = srtp_filter_.UnprotectRtp(data, len, &len); |
| 870 | if (!res) { |
| 871 | int seq_num = -1; |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 872 | uint32_t ssrc = 0; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 873 | GetRtpSeqNum(data, len, &seq_num); |
| 874 | GetRtpSsrc(data, len, &ssrc); |
| 875 | LOG(LS_ERROR) << "Failed to unprotect " << content_name_ |
| 876 | << " RTP packet: size=" << len |
| 877 | << ", seqnum=" << seq_num << ", SSRC=" << ssrc; |
| 878 | return; |
| 879 | } |
| 880 | } else { |
| 881 | res = srtp_filter_.UnprotectRtcp(data, len, &len); |
| 882 | if (!res) { |
| 883 | int type = -1; |
| 884 | GetRtcpType(data, len, &type); |
| 885 | LOG(LS_ERROR) << "Failed to unprotect " << content_name_ |
| 886 | << " RTCP packet: size=" << len << ", type=" << type; |
| 887 | return; |
| 888 | } |
| 889 | } |
| 890 | |
kwiberg@webrtc.org | eebcab5 | 2015-03-24 09:19:06 +0000 | [diff] [blame] | 891 | packet->SetSize(len); |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 892 | } else if (srtp_required_) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 893 | // Our session description indicates that SRTP is required, but we got a |
| 894 | // packet before our SRTP filter is active. This means either that |
| 895 | // a) we got SRTP packets before we received the SDES keys, in which case |
| 896 | // we can't decrypt it anyway, or |
| 897 | // b) we got SRTP packets before DTLS completed on both the RTP and RTCP |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 898 | // transports, so we haven't yet extracted keys, even if DTLS did |
| 899 | // complete on the transport that the packets are being sent on. It's |
| 900 | // really good practice to wait for both RTP and RTCP to be good to go |
| 901 | // before sending media, to prevent weird failure modes, so it's fine |
| 902 | // for us to just eat packets here. This is all sidestepped if RTCP mux |
| 903 | // is used anyway. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 904 | LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp) |
| 905 | << " packet when SRTP is inactive and crypto is required"; |
| 906 | return; |
| 907 | } |
| 908 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 909 | invoker_.AsyncInvoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 910 | RTC_FROM_HERE, worker_thread_, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 911 | Bind(&BaseChannel::OnPacketReceived, this, rtcp, *packet, packet_time)); |
| 912 | } |
| 913 | |
| 914 | void BaseChannel::OnPacketReceived(bool rtcp, |
| 915 | const rtc::CopyOnWriteBuffer& packet, |
| 916 | const rtc::PacketTime& packet_time) { |
| 917 | RTC_DCHECK(worker_thread_->IsCurrent()); |
| 918 | // Need to copy variable because OnRtcpReceived/OnPacketReceived |
| 919 | // requires non-const pointer to buffer. This doesn't memcpy the actual data. |
| 920 | rtc::CopyOnWriteBuffer data(packet); |
| 921 | if (rtcp) { |
| 922 | media_channel_->OnRtcpReceived(&data, packet_time); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 923 | } else { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 924 | media_channel_->OnPacketReceived(&data, packet_time); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 925 | } |
| 926 | } |
| 927 | |
pthatcher@webrtc.org | 592470b | 2015-03-16 21:15:37 +0000 | [diff] [blame] | 928 | bool BaseChannel::PushdownLocalDescription( |
| 929 | const SessionDescription* local_desc, ContentAction action, |
| 930 | std::string* error_desc) { |
| 931 | const ContentInfo* content_info = GetFirstContent(local_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 932 | const MediaContentDescription* content_desc = |
| 933 | GetContentDescription(content_info); |
| 934 | if (content_desc && content_info && !content_info->rejected && |
pthatcher@webrtc.org | 592470b | 2015-03-16 21:15:37 +0000 | [diff] [blame] | 935 | !SetLocalContent(content_desc, action, error_desc)) { |
| 936 | LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action; |
| 937 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 938 | } |
pthatcher@webrtc.org | 592470b | 2015-03-16 21:15:37 +0000 | [diff] [blame] | 939 | return true; |
| 940 | } |
| 941 | |
| 942 | bool BaseChannel::PushdownRemoteDescription( |
| 943 | const SessionDescription* remote_desc, ContentAction action, |
| 944 | std::string* error_desc) { |
| 945 | const ContentInfo* content_info = GetFirstContent(remote_desc); |
| 946 | const MediaContentDescription* content_desc = |
| 947 | GetContentDescription(content_info); |
| 948 | if (content_desc && content_info && !content_info->rejected && |
| 949 | !SetRemoteContent(content_desc, action, error_desc)) { |
| 950 | LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action; |
| 951 | return false; |
| 952 | } |
| 953 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 954 | } |
| 955 | |
| 956 | void BaseChannel::EnableMedia_w() { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 957 | RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 958 | if (enabled_) |
| 959 | return; |
| 960 | |
| 961 | LOG(LS_INFO) << "Channel enabled"; |
| 962 | enabled_ = true; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 963 | UpdateMediaSendRecvState_w(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 964 | } |
| 965 | |
| 966 | void BaseChannel::DisableMedia_w() { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 967 | RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 968 | if (!enabled_) |
| 969 | return; |
| 970 | |
| 971 | LOG(LS_INFO) << "Channel disabled"; |
| 972 | enabled_ = false; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 973 | UpdateMediaSendRecvState_w(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 974 | } |
| 975 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 976 | void BaseChannel::UpdateWritableState_n() { |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 977 | rtc::PacketTransportInternal* rtp_packet_transport = |
| 978 | rtp_transport_.rtp_packet_transport(); |
| 979 | rtc::PacketTransportInternal* rtcp_packet_transport = |
| 980 | rtp_transport_.rtcp_packet_transport(); |
| 981 | if (rtp_packet_transport && rtp_packet_transport->writable() && |
| 982 | (!rtcp_packet_transport || rtcp_packet_transport->writable())) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 983 | ChannelWritable_n(); |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 984 | } else { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 985 | ChannelNotWritable_n(); |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 986 | } |
| 987 | } |
| 988 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 989 | void BaseChannel::ChannelWritable_n() { |
| 990 | RTC_DCHECK(network_thread_->IsCurrent()); |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 991 | if (writable_) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 992 | return; |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 993 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 994 | |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 995 | LOG(LS_INFO) << "Channel writable (" << content_name_ << ")" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 996 | << (was_ever_writable_ ? "" : " for the first time"); |
| 997 | |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 998 | if (selected_candidate_pair_) |
| 999 | LOG(LS_INFO) |
| 1000 | << "Using " |
| 1001 | << selected_candidate_pair_->local_candidate().ToSensitiveString() |
| 1002 | << "->" |
| 1003 | << selected_candidate_pair_->remote_candidate().ToSensitiveString(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1004 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1005 | was_ever_writable_ = true; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1006 | MaybeSetupDtlsSrtp_n(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1007 | writable_ = true; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1008 | UpdateMediaSendRecvState(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1009 | } |
| 1010 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 1011 | void BaseChannel::SignalDtlsSrtpSetupFailure_n(bool rtcp) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1012 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 1013 | invoker_.AsyncInvoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1014 | RTC_FROM_HERE, signaling_thread(), |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 1015 | Bind(&BaseChannel::SignalDtlsSrtpSetupFailure_s, this, rtcp)); |
pthatcher@webrtc.org | 4eeef58 | 2015-03-16 19:34:23 +0000 | [diff] [blame] | 1016 | } |
| 1017 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 1018 | void BaseChannel::SignalDtlsSrtpSetupFailure_s(bool rtcp) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1019 | RTC_DCHECK(signaling_thread() == rtc::Thread::Current()); |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 1020 | SignalDtlsSrtpSetupFailure(this, rtcp); |
pthatcher@webrtc.org | 4eeef58 | 2015-03-16 19:34:23 +0000 | [diff] [blame] | 1021 | } |
| 1022 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1023 | bool BaseChannel::SetDtlsSrtpCryptoSuites_n(DtlsTransportInternal* transport, |
| 1024 | bool rtcp) { |
Guo-wei Shieh | 521ed7b | 2015-11-18 19:41:53 -0800 | [diff] [blame] | 1025 | std::vector<int> crypto_suites; |
| 1026 | // We always use the default SRTP crypto suites for RTCP, but we may use |
| 1027 | // different crypto suites for RTP depending on the media type. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1028 | if (!rtcp) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1029 | GetSrtpCryptoSuites_n(&crypto_suites); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1030 | } else { |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 1031 | GetDefaultSrtpCryptoSuites(crypto_options(), &crypto_suites); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1032 | } |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1033 | return transport->SetSrtpCryptoSuites(crypto_suites); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1034 | } |
| 1035 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1036 | bool BaseChannel::ShouldSetupDtlsSrtp_n() const { |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1037 | // Since DTLS is applied to all transports, checking RTP should be enough. |
| 1038 | return rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1039 | } |
| 1040 | |
| 1041 | // This function returns true if either DTLS-SRTP is not in use |
| 1042 | // *or* DTLS-SRTP is successfully set up. |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1043 | bool BaseChannel::SetupDtlsSrtp_n(bool rtcp) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1044 | RTC_DCHECK(network_thread_->IsCurrent()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1045 | bool ret = false; |
| 1046 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1047 | DtlsTransportInternal* transport = |
| 1048 | rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_; |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 1049 | RTC_DCHECK(transport); |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1050 | RTC_DCHECK(transport->IsDtlsActive()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1051 | |
Guo-wei Shieh | 521ed7b | 2015-11-18 19:41:53 -0800 | [diff] [blame] | 1052 | int selected_crypto_suite; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1053 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1054 | if (!transport->GetSrtpCryptoSuite(&selected_crypto_suite)) { |
Guo-wei Shieh | 521ed7b | 2015-11-18 19:41:53 -0800 | [diff] [blame] | 1055 | LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite"; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1056 | return false; |
| 1057 | } |
| 1058 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1059 | LOG(LS_INFO) << "Installing keys from DTLS-SRTP on " << content_name() << " " |
| 1060 | << PacketType(rtcp); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1061 | |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 1062 | int key_len; |
| 1063 | int salt_len; |
| 1064 | if (!rtc::GetSrtpKeyAndSaltLengths(selected_crypto_suite, &key_len, |
| 1065 | &salt_len)) { |
| 1066 | LOG(LS_ERROR) << "Unknown DTLS-SRTP crypto suite" << selected_crypto_suite; |
| 1067 | return false; |
| 1068 | } |
| 1069 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1070 | // OK, we're now doing DTLS (RFC 5764) |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 1071 | std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1072 | |
| 1073 | // RFC 5705 exporter using the RFC 5764 parameters |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1074 | if (!transport->ExportKeyingMaterial(kDtlsSrtpExporterLabel, NULL, 0, false, |
| 1075 | &dtls_buffer[0], dtls_buffer.size())) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1076 | LOG(LS_WARNING) << "DTLS-SRTP key export failed"; |
nisse | eb4ca4e | 2017-01-12 02:24:27 -0800 | [diff] [blame] | 1077 | RTC_NOTREACHED(); // This should never happen |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1078 | return false; |
| 1079 | } |
| 1080 | |
| 1081 | // Sync up the keys with the DTLS-SRTP interface |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 1082 | std::vector<unsigned char> client_write_key(key_len + salt_len); |
| 1083 | std::vector<unsigned char> server_write_key(key_len + salt_len); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1084 | size_t offset = 0; |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 1085 | memcpy(&client_write_key[0], &dtls_buffer[offset], key_len); |
| 1086 | offset += key_len; |
| 1087 | memcpy(&server_write_key[0], &dtls_buffer[offset], key_len); |
| 1088 | offset += key_len; |
| 1089 | memcpy(&client_write_key[key_len], &dtls_buffer[offset], salt_len); |
| 1090 | offset += salt_len; |
| 1091 | memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1092 | |
| 1093 | std::vector<unsigned char> *send_key, *recv_key; |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1094 | rtc::SSLRole role; |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1095 | if (!transport->GetSslRole(&role)) { |
sergeyu@chromium.org | 0be6aa0 | 2013-08-23 23:21:25 +0000 | [diff] [blame] | 1096 | LOG(LS_WARNING) << "GetSslRole failed"; |
| 1097 | return false; |
| 1098 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1099 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1100 | if (role == rtc::SSL_SERVER) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1101 | send_key = &server_write_key; |
| 1102 | recv_key = &client_write_key; |
| 1103 | } else { |
| 1104 | send_key = &client_write_key; |
| 1105 | recv_key = &server_write_key; |
| 1106 | } |
| 1107 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1108 | if (rtcp) { |
Guo-wei Shieh | 521ed7b | 2015-11-18 19:41:53 -0800 | [diff] [blame] | 1109 | ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0], |
| 1110 | static_cast<int>(send_key->size()), |
| 1111 | selected_crypto_suite, &(*recv_key)[0], |
| 1112 | static_cast<int>(recv_key->size())); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1113 | } else { |
Guo-wei Shieh | 521ed7b | 2015-11-18 19:41:53 -0800 | [diff] [blame] | 1114 | ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0], |
| 1115 | static_cast<int>(send_key->size()), |
| 1116 | selected_crypto_suite, &(*recv_key)[0], |
| 1117 | static_cast<int>(recv_key->size())); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1118 | } |
| 1119 | |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 1120 | if (!ret) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1121 | LOG(LS_WARNING) << "DTLS-SRTP key installation failed"; |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 1122 | } else { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1123 | dtls_keyed_ = true; |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 1124 | UpdateTransportOverhead(); |
| 1125 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1126 | return ret; |
| 1127 | } |
| 1128 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1129 | void BaseChannel::MaybeSetupDtlsSrtp_n() { |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 1130 | if (srtp_filter_.IsActive()) { |
| 1131 | return; |
| 1132 | } |
| 1133 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1134 | if (!ShouldSetupDtlsSrtp_n()) { |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 1135 | return; |
| 1136 | } |
| 1137 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1138 | if (!SetupDtlsSrtp_n(false)) { |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 1139 | SignalDtlsSrtpSetupFailure_n(false); |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 1140 | return; |
| 1141 | } |
| 1142 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1143 | if (rtcp_dtls_transport_) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1144 | if (!SetupDtlsSrtp_n(true)) { |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 1145 | SignalDtlsSrtpSetupFailure_n(true); |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 1146 | return; |
| 1147 | } |
| 1148 | } |
| 1149 | } |
| 1150 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1151 | void BaseChannel::ChannelNotWritable_n() { |
| 1152 | RTC_DCHECK(network_thread_->IsCurrent()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1153 | if (!writable_) |
| 1154 | return; |
| 1155 | |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1156 | LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")"; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1157 | writable_ = false; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1158 | UpdateMediaSendRecvState(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1159 | } |
| 1160 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1161 | bool BaseChannel::SetRtpTransportParameters( |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1162 | const MediaContentDescription* content, |
| 1163 | ContentAction action, |
| 1164 | ContentSource src, |
| 1165 | std::string* error_desc) { |
| 1166 | if (action == CA_UPDATE) { |
| 1167 | // These parameters never get changed by a CA_UDPATE. |
| 1168 | return true; |
| 1169 | } |
| 1170 | |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 1171 | // Cache srtp_required_ for belt and suspenders check on SendPacket |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1172 | return network_thread_->Invoke<bool>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1173 | RTC_FROM_HERE, Bind(&BaseChannel::SetRtpTransportParameters_n, this, |
| 1174 | content, action, src, error_desc)); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1175 | } |
| 1176 | |
| 1177 | bool BaseChannel::SetRtpTransportParameters_n( |
| 1178 | const MediaContentDescription* content, |
| 1179 | ContentAction action, |
| 1180 | ContentSource src, |
| 1181 | std::string* error_desc) { |
| 1182 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 1183 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1184 | if (!SetSrtp_n(content->cryptos(), action, src, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1185 | return false; |
| 1186 | } |
| 1187 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1188 | if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1189 | return false; |
| 1190 | } |
| 1191 | |
| 1192 | return true; |
| 1193 | } |
| 1194 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1195 | // |dtls| will be set to true if DTLS is active for transport and crypto is |
| 1196 | // empty. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1197 | bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos, |
| 1198 | bool* dtls, |
| 1199 | std::string* error_desc) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 1200 | *dtls = rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive(); |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1201 | if (*dtls && !cryptos.empty()) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1202 | SafeSetError("Cryptos must be empty when DTLS is active.", error_desc); |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1203 | return false; |
| 1204 | } |
| 1205 | return true; |
| 1206 | } |
| 1207 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1208 | bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1209 | ContentAction action, |
| 1210 | ContentSource src, |
| 1211 | std::string* error_desc) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1212 | TRACE_EVENT0("webrtc", "BaseChannel::SetSrtp_w"); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1213 | if (action == CA_UPDATE) { |
| 1214 | // no crypto params. |
| 1215 | return true; |
| 1216 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1217 | bool ret = false; |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1218 | bool dtls = false; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1219 | ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1220 | if (!ret) { |
| 1221 | return false; |
| 1222 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1223 | switch (action) { |
| 1224 | case CA_OFFER: |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1225 | // If DTLS is already active on the channel, we could be renegotiating |
| 1226 | // here. We don't update the srtp filter. |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1227 | if (!dtls) { |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 +0000 | [diff] [blame] | 1228 | ret = srtp_filter_.SetOffer(cryptos, src); |
| 1229 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1230 | break; |
| 1231 | case CA_PRANSWER: |
| 1232 | // If we're doing DTLS-SRTP, we don't want to update the filter |
| 1233 | // with an answer, because we already have SRTP parameters. |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1234 | if (!dtls) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1235 | ret = srtp_filter_.SetProvisionalAnswer(cryptos, src); |
| 1236 | } |
| 1237 | break; |
| 1238 | case CA_ANSWER: |
| 1239 | // If we're doing DTLS-SRTP, we don't want to update the filter |
| 1240 | // with an answer, because we already have SRTP parameters. |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1241 | if (!dtls) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1242 | ret = srtp_filter_.SetAnswer(cryptos, src); |
| 1243 | } |
| 1244 | break; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1245 | default: |
| 1246 | break; |
| 1247 | } |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1248 | if (!ret) { |
| 1249 | SafeSetError("Failed to setup SRTP filter.", error_desc); |
| 1250 | return false; |
| 1251 | } |
| 1252 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1253 | } |
| 1254 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1255 | bool BaseChannel::SetRtcpMux_n(bool enable, |
| 1256 | ContentAction action, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1257 | ContentSource src, |
| 1258 | std::string* error_desc) { |
deadbeef | 8e814d7 | 2017-01-13 11:34:39 -0800 | [diff] [blame] | 1259 | // Provide a more specific error message for the RTCP mux "require" policy |
| 1260 | // case. |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 1261 | if (rtp_transport_.rtcp_mux_required() && !enable) { |
deadbeef | 8e814d7 | 2017-01-13 11:34:39 -0800 | [diff] [blame] | 1262 | SafeSetError( |
| 1263 | "rtcpMuxPolicy is 'require', but media description does not " |
| 1264 | "contain 'a=rtcp-mux'.", |
| 1265 | error_desc); |
| 1266 | return false; |
| 1267 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1268 | bool ret = false; |
| 1269 | switch (action) { |
| 1270 | case CA_OFFER: |
| 1271 | ret = rtcp_mux_filter_.SetOffer(enable, src); |
| 1272 | break; |
| 1273 | case CA_PRANSWER: |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1274 | // This may activate RTCP muxing, but we don't yet destroy the transport |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1275 | // because the final answer may deactivate it. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1276 | ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src); |
| 1277 | break; |
| 1278 | case CA_ANSWER: |
| 1279 | ret = rtcp_mux_filter_.SetAnswer(enable, src); |
| 1280 | if (ret && rtcp_mux_filter_.IsActive()) { |
deadbeef | e814a0d | 2017-02-25 18:15:09 -0800 | [diff] [blame] | 1281 | // We permanently activated RTCP muxing; signal that we no longer need |
| 1282 | // the RTCP transport. |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 1283 | std::string debug_name = |
| 1284 | transport_name_.empty() |
| 1285 | ? rtp_transport_.rtp_packet_transport()->debug_name() |
| 1286 | : transport_name_; |
deadbeef | e814a0d | 2017-02-25 18:15:09 -0800 | [diff] [blame] | 1287 | ; |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1288 | LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name() |
deadbeef | e814a0d | 2017-02-25 18:15:09 -0800 | [diff] [blame] | 1289 | << "; no longer need RTCP transport for " << debug_name; |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 1290 | if (rtp_transport_.rtcp_packet_transport()) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 1291 | SetTransport_n(true, nullptr, nullptr); |
| 1292 | SignalRtcpMuxFullyActive(transport_name_); |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 1293 | } |
deadbeef | 062ce9f | 2016-08-26 21:42:15 -0700 | [diff] [blame] | 1294 | UpdateWritableState_n(); |
| 1295 | SetTransportChannelReadyToSend(true, false); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1296 | } |
| 1297 | break; |
| 1298 | case CA_UPDATE: |
| 1299 | // No RTCP mux info. |
| 1300 | ret = true; |
Henrik Kjellander | 7c027b6 | 2015-04-22 13:21:30 +0200 | [diff] [blame] | 1301 | break; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1302 | default: |
| 1303 | break; |
| 1304 | } |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1305 | if (!ret) { |
| 1306 | SafeSetError("Failed to setup RTCP mux filter.", error_desc); |
| 1307 | return false; |
| 1308 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1309 | // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 1310 | // CA_ANSWER, but we only want to tear down the RTCP transport if we received |
| 1311 | // a final answer. |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1312 | if (rtcp_mux_filter_.IsActive()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1313 | // If the RTP transport is already writable, then so are we. |
zstein | d48dbda | 2017-04-04 19:45:57 -0700 | [diff] [blame^] | 1314 | if (rtp_transport_.rtp_packet_transport()->writable()) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1315 | ChannelWritable_n(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1316 | } |
| 1317 | } |
| 1318 | |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1319 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1320 | } |
| 1321 | |
| 1322 | bool BaseChannel::AddRecvStream_w(const StreamParams& sp) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1323 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
pbos | 482b12e | 2015-11-16 10:19:58 -0800 | [diff] [blame] | 1324 | return media_channel()->AddRecvStream(sp); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1325 | } |
| 1326 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1327 | bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1328 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1329 | return media_channel()->RemoveRecvStream(ssrc); |
| 1330 | } |
| 1331 | |
| 1332 | bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1333 | ContentAction action, |
| 1334 | std::string* error_desc) { |
nisse | 7ce109a | 2017-01-31 00:57:56 -0800 | [diff] [blame] | 1335 | if (!(action == CA_OFFER || action == CA_ANSWER || |
| 1336 | action == CA_PRANSWER || action == CA_UPDATE)) |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1337 | return false; |
| 1338 | |
| 1339 | // If this is an update, streams only contain streams that have changed. |
| 1340 | if (action == CA_UPDATE) { |
| 1341 | for (StreamParamsVec::const_iterator it = streams.begin(); |
| 1342 | it != streams.end(); ++it) { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1343 | const StreamParams* existing_stream = |
| 1344 | GetStreamByIds(local_streams_, it->groupid, it->id); |
| 1345 | if (!existing_stream && it->has_ssrcs()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1346 | if (media_channel()->AddSendStream(*it)) { |
| 1347 | local_streams_.push_back(*it); |
| 1348 | LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc(); |
| 1349 | } else { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1350 | std::ostringstream desc; |
| 1351 | desc << "Failed to add send stream ssrc: " << it->first_ssrc(); |
| 1352 | SafeSetError(desc.str(), error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1353 | return false; |
| 1354 | } |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1355 | } else if (existing_stream && !it->has_ssrcs()) { |
| 1356 | if (!media_channel()->RemoveSendStream(existing_stream->first_ssrc())) { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1357 | std::ostringstream desc; |
| 1358 | desc << "Failed to remove send stream with ssrc " |
| 1359 | << it->first_ssrc() << "."; |
| 1360 | SafeSetError(desc.str(), error_desc); |
| 1361 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1362 | } |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1363 | RemoveStreamBySsrc(&local_streams_, existing_stream->first_ssrc()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1364 | } else { |
| 1365 | LOG(LS_WARNING) << "Ignore unsupported stream update"; |
| 1366 | } |
| 1367 | } |
| 1368 | return true; |
| 1369 | } |
| 1370 | // Else streams are all the streams we want to send. |
| 1371 | |
| 1372 | // Check for streams that have been removed. |
| 1373 | bool ret = true; |
| 1374 | for (StreamParamsVec::const_iterator it = local_streams_.begin(); |
| 1375 | it != local_streams_.end(); ++it) { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1376 | if (!GetStreamBySsrc(streams, it->first_ssrc())) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1377 | if (!media_channel()->RemoveSendStream(it->first_ssrc())) { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1378 | std::ostringstream desc; |
| 1379 | desc << "Failed to remove send stream with ssrc " |
| 1380 | << it->first_ssrc() << "."; |
| 1381 | SafeSetError(desc.str(), error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1382 | ret = false; |
| 1383 | } |
| 1384 | } |
| 1385 | } |
| 1386 | // Check for new streams. |
| 1387 | for (StreamParamsVec::const_iterator it = streams.begin(); |
| 1388 | it != streams.end(); ++it) { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1389 | if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1390 | if (media_channel()->AddSendStream(*it)) { |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 1391 | LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0]; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1392 | } else { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1393 | std::ostringstream desc; |
| 1394 | desc << "Failed to add send stream ssrc: " << it->first_ssrc(); |
| 1395 | SafeSetError(desc.str(), error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1396 | ret = false; |
| 1397 | } |
| 1398 | } |
| 1399 | } |
| 1400 | local_streams_ = streams; |
| 1401 | return ret; |
| 1402 | } |
| 1403 | |
| 1404 | bool BaseChannel::UpdateRemoteStreams_w( |
| 1405 | const std::vector<StreamParams>& streams, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1406 | ContentAction action, |
| 1407 | std::string* error_desc) { |
nisse | 7ce109a | 2017-01-31 00:57:56 -0800 | [diff] [blame] | 1408 | if (!(action == CA_OFFER || action == CA_ANSWER || |
| 1409 | action == CA_PRANSWER || action == CA_UPDATE)) |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1410 | return false; |
| 1411 | |
| 1412 | // If this is an update, streams only contain streams that have changed. |
| 1413 | if (action == CA_UPDATE) { |
| 1414 | for (StreamParamsVec::const_iterator it = streams.begin(); |
| 1415 | it != streams.end(); ++it) { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1416 | const StreamParams* existing_stream = |
| 1417 | GetStreamByIds(remote_streams_, it->groupid, it->id); |
| 1418 | if (!existing_stream && it->has_ssrcs()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1419 | if (AddRecvStream_w(*it)) { |
| 1420 | remote_streams_.push_back(*it); |
| 1421 | LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc(); |
| 1422 | } else { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1423 | std::ostringstream desc; |
| 1424 | desc << "Failed to add remote stream ssrc: " << it->first_ssrc(); |
| 1425 | SafeSetError(desc.str(), error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1426 | return false; |
| 1427 | } |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1428 | } else if (existing_stream && !it->has_ssrcs()) { |
| 1429 | if (!RemoveRecvStream_w(existing_stream->first_ssrc())) { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1430 | std::ostringstream desc; |
| 1431 | desc << "Failed to remove remote stream with ssrc " |
| 1432 | << it->first_ssrc() << "."; |
| 1433 | SafeSetError(desc.str(), error_desc); |
| 1434 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1435 | } |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1436 | RemoveStreamBySsrc(&remote_streams_, existing_stream->first_ssrc()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1437 | } else { |
| 1438 | LOG(LS_WARNING) << "Ignore unsupported stream update." |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1439 | << " Stream exists? " << (existing_stream != nullptr) |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1440 | << " new stream = " << it->ToString(); |
| 1441 | } |
| 1442 | } |
| 1443 | return true; |
| 1444 | } |
| 1445 | // Else streams are all the streams we want to receive. |
| 1446 | |
| 1447 | // Check for streams that have been removed. |
| 1448 | bool ret = true; |
| 1449 | for (StreamParamsVec::const_iterator it = remote_streams_.begin(); |
| 1450 | it != remote_streams_.end(); ++it) { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1451 | if (!GetStreamBySsrc(streams, it->first_ssrc())) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1452 | if (!RemoveRecvStream_w(it->first_ssrc())) { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1453 | std::ostringstream desc; |
| 1454 | desc << "Failed to remove remote stream with ssrc " |
| 1455 | << it->first_ssrc() << "."; |
| 1456 | SafeSetError(desc.str(), error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1457 | ret = false; |
| 1458 | } |
| 1459 | } |
| 1460 | } |
| 1461 | // Check for new streams. |
| 1462 | for (StreamParamsVec::const_iterator it = streams.begin(); |
| 1463 | it != streams.end(); ++it) { |
tommi@webrtc.org | 586f2ed | 2015-01-22 23:00:41 +0000 | [diff] [blame] | 1464 | if (!GetStreamBySsrc(remote_streams_, it->first_ssrc())) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1465 | if (AddRecvStream_w(*it)) { |
| 1466 | LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0]; |
| 1467 | } else { |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1468 | std::ostringstream desc; |
| 1469 | desc << "Failed to add remote stream ssrc: " << it->first_ssrc(); |
| 1470 | SafeSetError(desc.str(), error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1471 | ret = false; |
| 1472 | } |
| 1473 | } |
| 1474 | } |
| 1475 | remote_streams_ = streams; |
| 1476 | return ret; |
| 1477 | } |
| 1478 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1479 | void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w( |
isheriff | 6f8d686 | 2016-05-26 11:24:55 -0700 | [diff] [blame] | 1480 | const std::vector<webrtc::RtpExtension>& extensions) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1481 | // Absolute Send Time extension id is used only with external auth, |
| 1482 | // so do not bother searching for it and making asyncronious call to set |
| 1483 | // something that is not used. |
| 1484 | #if defined(ENABLE_EXTERNAL_AUTH) |
isheriff | 6f8d686 | 2016-05-26 11:24:55 -0700 | [diff] [blame] | 1485 | const webrtc::RtpExtension* send_time_extension = |
| 1486 | FindHeaderExtension(extensions, webrtc::RtpExtension::kAbsSendTimeUri); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1487 | int rtp_abs_sendtime_extn_id = |
henrike@webrtc.org | d43aa9d | 2014-02-21 23:43:24 +0000 | [diff] [blame] | 1488 | send_time_extension ? send_time_extension->id : -1; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1489 | invoker_.AsyncInvoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1490 | RTC_FROM_HERE, network_thread_, |
| 1491 | Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, this, |
| 1492 | rtp_abs_sendtime_extn_id)); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1493 | #endif |
| 1494 | } |
| 1495 | |
| 1496 | void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n( |
| 1497 | int rtp_abs_sendtime_extn_id) { |
| 1498 | rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id; |
henrike@webrtc.org | d43aa9d | 2014-02-21 23:43:24 +0000 | [diff] [blame] | 1499 | } |
| 1500 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1501 | void BaseChannel::OnMessage(rtc::Message *pmsg) { |
Peter Boström | 6f28cf0 | 2015-12-07 23:17:15 +0100 | [diff] [blame] | 1502 | TRACE_EVENT0("webrtc", "BaseChannel::OnMessage"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1503 | switch (pmsg->message_id) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1504 | case MSG_SEND_RTP_PACKET: |
| 1505 | case MSG_SEND_RTCP_PACKET: { |
| 1506 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 1507 | SendPacketMessageData* data = |
| 1508 | static_cast<SendPacketMessageData*>(pmsg->pdata); |
| 1509 | bool rtcp = pmsg->message_id == MSG_SEND_RTCP_PACKET; |
| 1510 | SendPacket(rtcp, &data->packet, data->options); |
| 1511 | delete data; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1512 | break; |
| 1513 | } |
| 1514 | case MSG_FIRSTPACKETRECEIVED: { |
| 1515 | SignalFirstPacketReceived(this); |
| 1516 | break; |
| 1517 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1518 | } |
| 1519 | } |
| 1520 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1521 | void BaseChannel::FlushRtcpMessages_n() { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1522 | // Flush all remaining RTCP messages. This should only be called in |
| 1523 | // destructor. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1524 | RTC_DCHECK(network_thread_->IsCurrent()); |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1525 | rtc::MessageList rtcp_messages; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1526 | network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages); |
| 1527 | for (const auto& message : rtcp_messages) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1528 | network_thread_->Send(RTC_FROM_HERE, this, MSG_SEND_RTCP_PACKET, |
| 1529 | message.pdata); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1530 | } |
| 1531 | } |
| 1532 | |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 1533 | void BaseChannel::SignalSentPacket_n( |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 1534 | rtc::PacketTransportInternal* /* transport */, |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 1535 | const rtc::SentPacket& sent_packet) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1536 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 1537 | invoker_.AsyncInvoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1538 | RTC_FROM_HERE, worker_thread_, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1539 | rtc::Bind(&BaseChannel::SignalSentPacket_w, this, sent_packet)); |
| 1540 | } |
| 1541 | |
| 1542 | void BaseChannel::SignalSentPacket_w(const rtc::SentPacket& sent_packet) { |
| 1543 | RTC_DCHECK(worker_thread_->IsCurrent()); |
| 1544 | SignalSentPacket(sent_packet); |
| 1545 | } |
| 1546 | |
| 1547 | VoiceChannel::VoiceChannel(rtc::Thread* worker_thread, |
| 1548 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 1549 | rtc::Thread* signaling_thread, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1550 | MediaEngineInterface* media_engine, |
| 1551 | VoiceMediaChannel* media_channel, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1552 | const std::string& content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 1553 | bool rtcp_mux_required, |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 1554 | bool srtp_required) |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1555 | : BaseChannel(worker_thread, |
| 1556 | network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 1557 | signaling_thread, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1558 | media_channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1559 | content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 1560 | rtcp_mux_required, |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 1561 | srtp_required), |
Fredrik Solenberg | 0c02264 | 2015-08-05 12:25:22 +0200 | [diff] [blame] | 1562 | media_engine_(media_engine), |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1563 | received_media_(false) {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1564 | |
| 1565 | VoiceChannel::~VoiceChannel() { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1566 | TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1567 | StopAudioMonitor(); |
| 1568 | StopMediaMonitor(); |
| 1569 | // this can't be done in the base class, since it calls a virtual |
| 1570 | DisableMedia_w(); |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 1571 | Deinit(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1572 | } |
| 1573 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1574 | bool VoiceChannel::SetAudioSend(uint32_t ssrc, |
solenberg | dfc8f4f | 2015-10-01 02:31:10 -0700 | [diff] [blame] | 1575 | bool enable, |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1576 | const AudioOptions* options, |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1577 | AudioSource* source) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1578 | return InvokeOnWorker(RTC_FROM_HERE, |
| 1579 | Bind(&VoiceMediaChannel::SetAudioSend, media_channel(), |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1580 | ssrc, enable, options, source)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1581 | } |
| 1582 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1583 | // TODO(juberti): Handle early media the right way. We should get an explicit |
| 1584 | // ringing message telling us to start playing local ringback, which we cancel |
| 1585 | // if any early media actually arrives. For now, we do the opposite, which is |
| 1586 | // to wait 1 second for early media, and start playing local ringback if none |
| 1587 | // arrives. |
| 1588 | void VoiceChannel::SetEarlyMedia(bool enable) { |
| 1589 | if (enable) { |
| 1590 | // Start the early media timeout |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1591 | worker_thread()->PostDelayed(RTC_FROM_HERE, kEarlyMediaTimeout, this, |
| 1592 | MSG_EARLYMEDIATIMEOUT); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1593 | } else { |
| 1594 | // Stop the timeout if currently going. |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 1595 | worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1596 | } |
| 1597 | } |
| 1598 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1599 | bool VoiceChannel::CanInsertDtmf() { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1600 | return InvokeOnWorker( |
| 1601 | RTC_FROM_HERE, Bind(&VoiceMediaChannel::CanInsertDtmf, media_channel())); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1602 | } |
| 1603 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1604 | bool VoiceChannel::InsertDtmf(uint32_t ssrc, |
| 1605 | int event_code, |
solenberg | 1d63dd0 | 2015-12-02 12:35:09 -0800 | [diff] [blame] | 1606 | int duration) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1607 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceChannel::InsertDtmf_w, this, |
| 1608 | ssrc, event_code, duration)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1609 | } |
| 1610 | |
solenberg | 4bac9c5 | 2015-10-09 02:32:53 -0700 | [diff] [blame] | 1611 | bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1612 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetOutputVolume, |
| 1613 | media_channel(), ssrc, volume)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1614 | } |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 1615 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 1616 | void VoiceChannel::SetRawAudioSink( |
| 1617 | uint32_t ssrc, |
kwiberg | 3102294 | 2016-03-11 14:18:21 -0800 | [diff] [blame] | 1618 | std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
| 1619 | // We need to work around Bind's lack of support for unique_ptr and ownership |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 1620 | // passing. So we invoke to our own little routine that gets a pointer to |
| 1621 | // our local variable. This is OK since we're synchronously invoking. |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1622 | InvokeOnWorker(RTC_FROM_HERE, |
| 1623 | Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink)); |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 1624 | } |
| 1625 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1626 | webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const { |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1627 | return worker_thread()->Invoke<webrtc::RtpParameters>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1628 | RTC_FROM_HERE, Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc)); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1629 | } |
| 1630 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1631 | webrtc::RtpParameters VoiceChannel::GetRtpSendParameters_w( |
| 1632 | uint32_t ssrc) const { |
| 1633 | return media_channel()->GetRtpSendParameters(ssrc); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1634 | } |
| 1635 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1636 | bool VoiceChannel::SetRtpSendParameters( |
| 1637 | uint32_t ssrc, |
| 1638 | const webrtc::RtpParameters& parameters) { |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1639 | return InvokeOnWorker( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1640 | RTC_FROM_HERE, |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1641 | Bind(&VoiceChannel::SetRtpSendParameters_w, this, ssrc, parameters)); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1642 | } |
| 1643 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1644 | bool VoiceChannel::SetRtpSendParameters_w(uint32_t ssrc, |
| 1645 | webrtc::RtpParameters parameters) { |
| 1646 | return media_channel()->SetRtpSendParameters(ssrc, parameters); |
| 1647 | } |
| 1648 | |
| 1649 | webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters( |
| 1650 | uint32_t ssrc) const { |
| 1651 | return worker_thread()->Invoke<webrtc::RtpParameters>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1652 | RTC_FROM_HERE, |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1653 | Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc)); |
| 1654 | } |
| 1655 | |
| 1656 | webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w( |
| 1657 | uint32_t ssrc) const { |
| 1658 | return media_channel()->GetRtpReceiveParameters(ssrc); |
| 1659 | } |
| 1660 | |
| 1661 | bool VoiceChannel::SetRtpReceiveParameters( |
| 1662 | uint32_t ssrc, |
| 1663 | const webrtc::RtpParameters& parameters) { |
| 1664 | return InvokeOnWorker( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1665 | RTC_FROM_HERE, |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1666 | Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters)); |
| 1667 | } |
| 1668 | |
| 1669 | bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc, |
| 1670 | webrtc::RtpParameters parameters) { |
| 1671 | return media_channel()->SetRtpReceiveParameters(ssrc, parameters); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1672 | } |
| 1673 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1674 | bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1675 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats, |
| 1676 | media_channel(), stats)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1677 | } |
| 1678 | |
| 1679 | void VoiceChannel::StartMediaMonitor(int cms) { |
| 1680 | media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1681 | rtc::Thread::Current())); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1682 | media_monitor_->SignalUpdate.connect( |
| 1683 | this, &VoiceChannel::OnMediaMonitorUpdate); |
| 1684 | media_monitor_->Start(cms); |
| 1685 | } |
| 1686 | |
| 1687 | void VoiceChannel::StopMediaMonitor() { |
| 1688 | if (media_monitor_) { |
| 1689 | media_monitor_->Stop(); |
| 1690 | media_monitor_->SignalUpdate.disconnect(this); |
| 1691 | media_monitor_.reset(); |
| 1692 | } |
| 1693 | } |
| 1694 | |
| 1695 | void VoiceChannel::StartAudioMonitor(int cms) { |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1696 | audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current())); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1697 | audio_monitor_ |
| 1698 | ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate); |
| 1699 | audio_monitor_->Start(cms); |
| 1700 | } |
| 1701 | |
| 1702 | void VoiceChannel::StopAudioMonitor() { |
| 1703 | if (audio_monitor_) { |
| 1704 | audio_monitor_->Stop(); |
| 1705 | audio_monitor_.reset(); |
| 1706 | } |
| 1707 | } |
| 1708 | |
| 1709 | bool VoiceChannel::IsAudioMonitorRunning() const { |
| 1710 | return (audio_monitor_.get() != NULL); |
| 1711 | } |
| 1712 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1713 | int VoiceChannel::GetInputLevel_w() { |
Fredrik Solenberg | 0c02264 | 2015-08-05 12:25:22 +0200 | [diff] [blame] | 1714 | return media_engine_->GetInputLevel(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1715 | } |
| 1716 | |
| 1717 | int VoiceChannel::GetOutputLevel_w() { |
| 1718 | return media_channel()->GetOutputLevel(); |
| 1719 | } |
| 1720 | |
| 1721 | void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) { |
| 1722 | media_channel()->GetActiveStreams(actives); |
| 1723 | } |
| 1724 | |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 1725 | void VoiceChannel::OnPacketRead(rtc::PacketTransportInternal* transport, |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 1726 | const char* data, |
| 1727 | size_t len, |
| 1728 | const rtc::PacketTime& packet_time, |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 +0000 | [diff] [blame] | 1729 | int flags) { |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 1730 | BaseChannel::OnPacketRead(transport, data, len, packet_time, flags); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1731 | // Set a flag when we've received an RTP packet. If we're waiting for early |
| 1732 | // media, this will disable the timeout. |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 1733 | if (!received_media_ && !PacketIsRtcp(transport, data, len)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1734 | received_media_ = true; |
| 1735 | } |
| 1736 | } |
| 1737 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1738 | void BaseChannel::UpdateMediaSendRecvState() { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1739 | RTC_DCHECK(network_thread_->IsCurrent()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1740 | invoker_.AsyncInvoke<void>( |
| 1741 | RTC_FROM_HERE, worker_thread_, |
| 1742 | Bind(&BaseChannel::UpdateMediaSendRecvState_w, this)); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1743 | } |
| 1744 | |
michaelt | 79e0588 | 2016-11-08 02:50:09 -0800 | [diff] [blame] | 1745 | int BaseChannel::GetTransportOverheadPerPacket() const { |
| 1746 | RTC_DCHECK(network_thread_->IsCurrent()); |
| 1747 | |
| 1748 | if (!selected_candidate_pair_) |
| 1749 | return 0; |
| 1750 | |
| 1751 | int transport_overhead_per_packet = 0; |
| 1752 | |
| 1753 | constexpr int kIpv4Overhaed = 20; |
| 1754 | constexpr int kIpv6Overhaed = 40; |
| 1755 | transport_overhead_per_packet += |
| 1756 | selected_candidate_pair_->local_candidate().address().family() == AF_INET |
| 1757 | ? kIpv4Overhaed |
| 1758 | : kIpv6Overhaed; |
| 1759 | |
| 1760 | constexpr int kUdpOverhaed = 8; |
| 1761 | constexpr int kTcpOverhaed = 20; |
| 1762 | transport_overhead_per_packet += |
| 1763 | selected_candidate_pair_->local_candidate().protocol() == |
| 1764 | TCP_PROTOCOL_NAME |
| 1765 | ? kTcpOverhaed |
| 1766 | : kUdpOverhaed; |
| 1767 | |
| 1768 | if (secure()) { |
| 1769 | int srtp_overhead = 0; |
| 1770 | if (srtp_filter_.GetSrtpOverhead(&srtp_overhead)) |
| 1771 | transport_overhead_per_packet += srtp_overhead; |
| 1772 | } |
| 1773 | |
| 1774 | return transport_overhead_per_packet; |
| 1775 | } |
| 1776 | |
| 1777 | void BaseChannel::UpdateTransportOverhead() { |
| 1778 | int transport_overhead_per_packet = GetTransportOverheadPerPacket(); |
| 1779 | if (transport_overhead_per_packet) |
| 1780 | invoker_.AsyncInvoke<void>( |
| 1781 | RTC_FROM_HERE, worker_thread_, |
| 1782 | Bind(&MediaChannel::OnTransportOverheadChanged, media_channel_, |
| 1783 | transport_overhead_per_packet)); |
| 1784 | } |
| 1785 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1786 | void VoiceChannel::UpdateMediaSendRecvState_w() { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1787 | // Render incoming data if we're the active call, and we have the local |
| 1788 | // content. We receive data on the default channel and multiplexed streams. |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1789 | bool recv = IsReadyToReceiveMedia_w(); |
solenberg | 5b14b42 | 2015-10-01 04:10:31 -0700 | [diff] [blame] | 1790 | media_channel()->SetPlayout(recv); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1791 | |
| 1792 | // Send outgoing data if we're the active call, we have the remote content, |
| 1793 | // and we have had some form of connectivity. |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1794 | bool send = IsReadyToSendMedia_w(); |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1795 | media_channel()->SetSend(send); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1796 | |
| 1797 | LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send; |
| 1798 | } |
| 1799 | |
| 1800 | const ContentInfo* VoiceChannel::GetFirstContent( |
| 1801 | const SessionDescription* sdesc) { |
| 1802 | return GetFirstAudioContent(sdesc); |
| 1803 | } |
| 1804 | |
| 1805 | bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1806 | ContentAction action, |
| 1807 | std::string* error_desc) { |
Peter Boström | 9f45a45 | 2015-12-08 13:25:57 +0100 | [diff] [blame] | 1808 | TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1809 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1810 | LOG(LS_INFO) << "Setting local voice description"; |
| 1811 | |
| 1812 | const AudioContentDescription* audio = |
| 1813 | static_cast<const AudioContentDescription*>(content); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1814 | RTC_DCHECK(audio != NULL); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1815 | if (!audio) { |
| 1816 | SafeSetError("Can't find audio content in local description.", error_desc); |
| 1817 | return false; |
| 1818 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1819 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1820 | if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1821 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1822 | } |
| 1823 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1824 | AudioRecvParameters recv_params = last_recv_params_; |
| 1825 | RtpParametersFromMediaDescription(audio, &recv_params); |
| 1826 | if (!media_channel()->SetRecvParameters(recv_params)) { |
Peter Thatcher | bfab5cb | 2015-08-20 17:40:24 -0700 | [diff] [blame] | 1827 | SafeSetError("Failed to set local audio description recv parameters.", |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1828 | error_desc); |
| 1829 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1830 | } |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1831 | for (const AudioCodec& codec : audio->codecs()) { |
| 1832 | bundle_filter()->AddPayloadType(codec.id); |
| 1833 | } |
| 1834 | last_recv_params_ = recv_params; |
| 1835 | |
| 1836 | // TODO(pthatcher): Move local streams into AudioSendParameters, and |
| 1837 | // only give it to the media channel once we have a remote |
| 1838 | // description too (without a remote description, we won't be able |
| 1839 | // to send them anyway). |
| 1840 | if (!UpdateLocalStreams_w(audio->streams(), action, error_desc)) { |
| 1841 | SafeSetError("Failed to set local audio description streams.", error_desc); |
| 1842 | return false; |
| 1843 | } |
| 1844 | |
| 1845 | set_local_content_direction(content->direction()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1846 | UpdateMediaSendRecvState_w(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1847 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1848 | } |
| 1849 | |
| 1850 | bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1851 | ContentAction action, |
| 1852 | std::string* error_desc) { |
Peter Boström | 9f45a45 | 2015-12-08 13:25:57 +0100 | [diff] [blame] | 1853 | TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1854 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1855 | LOG(LS_INFO) << "Setting remote voice description"; |
| 1856 | |
| 1857 | const AudioContentDescription* audio = |
| 1858 | static_cast<const AudioContentDescription*>(content); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1859 | RTC_DCHECK(audio != NULL); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 1860 | if (!audio) { |
| 1861 | SafeSetError("Can't find audio content in remote description.", error_desc); |
| 1862 | return false; |
| 1863 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1864 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1865 | if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1866 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1867 | } |
| 1868 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1869 | AudioSendParameters send_params = last_send_params_; |
| 1870 | RtpSendParametersFromMediaDescription(audio, &send_params); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1871 | if (audio->agc_minus_10db()) { |
Karl Wiberg | be57983 | 2015-11-10 22:34:18 +0100 | [diff] [blame] | 1872 | send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1873 | } |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1874 | |
| 1875 | bool parameters_applied = media_channel()->SetSendParameters(send_params); |
| 1876 | if (!parameters_applied) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1877 | SafeSetError("Failed to set remote audio description send parameters.", |
| 1878 | error_desc); |
| 1879 | return false; |
| 1880 | } |
| 1881 | last_send_params_ = send_params; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1882 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1883 | // TODO(pthatcher): Move remote streams into AudioRecvParameters, |
| 1884 | // and only give it to the media channel once we have a local |
| 1885 | // description too (without a local description, we won't be able to |
| 1886 | // recv them anyway). |
| 1887 | if (!UpdateRemoteStreams_w(audio->streams(), action, error_desc)) { |
| 1888 | SafeSetError("Failed to set remote audio description streams.", error_desc); |
| 1889 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1890 | } |
| 1891 | |
Peter Thatcher | bfab5cb | 2015-08-20 17:40:24 -0700 | [diff] [blame] | 1892 | if (audio->rtp_header_extensions_set()) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1893 | MaybeCacheRtpAbsSendTimeHeaderExtension_w(audio->rtp_header_extensions()); |
Peter Thatcher | bfab5cb | 2015-08-20 17:40:24 -0700 | [diff] [blame] | 1894 | } |
| 1895 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1896 | set_remote_content_direction(content->direction()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1897 | UpdateMediaSendRecvState_w(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1898 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1899 | } |
| 1900 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1901 | void VoiceChannel::HandleEarlyMediaTimeout() { |
| 1902 | // This occurs on the main thread, not the worker thread. |
| 1903 | if (!received_media_) { |
| 1904 | LOG(LS_INFO) << "No early media received before timeout"; |
| 1905 | SignalEarlyMediaTimeout(this); |
| 1906 | } |
| 1907 | } |
| 1908 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1909 | bool VoiceChannel::InsertDtmf_w(uint32_t ssrc, |
| 1910 | int event, |
solenberg | 1d63dd0 | 2015-12-02 12:35:09 -0800 | [diff] [blame] | 1911 | int duration) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1912 | if (!enabled()) { |
| 1913 | return false; |
| 1914 | } |
solenberg | 1d63dd0 | 2015-12-02 12:35:09 -0800 | [diff] [blame] | 1915 | return media_channel()->InsertDtmf(ssrc, event, duration); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1916 | } |
| 1917 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 1918 | void VoiceChannel::OnMessage(rtc::Message *pmsg) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1919 | switch (pmsg->message_id) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1920 | case MSG_EARLYMEDIATIMEOUT: |
| 1921 | HandleEarlyMediaTimeout(); |
| 1922 | break; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1923 | case MSG_CHANNEL_ERROR: { |
| 1924 | VoiceChannelErrorMessageData* data = |
| 1925 | static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1926 | delete data; |
| 1927 | break; |
| 1928 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1929 | default: |
| 1930 | BaseChannel::OnMessage(pmsg); |
| 1931 | break; |
| 1932 | } |
| 1933 | } |
| 1934 | |
| 1935 | void VoiceChannel::OnConnectionMonitorUpdate( |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 1936 | ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1937 | SignalConnectionMonitor(this, infos); |
| 1938 | } |
| 1939 | |
| 1940 | void VoiceChannel::OnMediaMonitorUpdate( |
| 1941 | VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 1942 | RTC_DCHECK(media_channel == this->media_channel()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1943 | SignalMediaMonitor(this, info); |
| 1944 | } |
| 1945 | |
| 1946 | void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor, |
| 1947 | const AudioInfo& info) { |
| 1948 | SignalAudioMonitor(this, info); |
| 1949 | } |
| 1950 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1951 | void VoiceChannel::GetSrtpCryptoSuites_n( |
| 1952 | std::vector<int>* crypto_suites) const { |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 1953 | GetSupportedAudioCryptoSuites(crypto_options(), crypto_suites); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1954 | } |
| 1955 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1956 | VideoChannel::VideoChannel(rtc::Thread* worker_thread, |
| 1957 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 1958 | rtc::Thread* signaling_thread, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1959 | VideoMediaChannel* media_channel, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1960 | const std::string& content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 1961 | bool rtcp_mux_required, |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 1962 | bool srtp_required) |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 1963 | : BaseChannel(worker_thread, |
| 1964 | network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 1965 | signaling_thread, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1966 | media_channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 1967 | content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 1968 | rtcp_mux_required, |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 1969 | srtp_required) {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1970 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1971 | VideoChannel::~VideoChannel() { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1972 | TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1973 | StopMediaMonitor(); |
| 1974 | // this can't be done in the base class, since it calls a virtual |
| 1975 | DisableMedia_w(); |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 1976 | |
| 1977 | Deinit(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1978 | } |
| 1979 | |
nisse | 08582ff | 2016-02-04 01:24:52 -0800 | [diff] [blame] | 1980 | bool VideoChannel::SetSink(uint32_t ssrc, |
nisse | acd935b | 2016-11-11 03:55:13 -0800 | [diff] [blame] | 1981 | rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
nisse | 08582ff | 2016-02-04 01:24:52 -0800 | [diff] [blame] | 1982 | worker_thread()->Invoke<void>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1983 | RTC_FROM_HERE, |
nisse | 08582ff | 2016-02-04 01:24:52 -0800 | [diff] [blame] | 1984 | Bind(&VideoMediaChannel::SetSink, media_channel(), ssrc, sink)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1985 | return true; |
| 1986 | } |
| 1987 | |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 1988 | bool VideoChannel::SetVideoSend( |
nisse | 2ded9b1 | 2016-04-08 02:23:55 -0700 | [diff] [blame] | 1989 | uint32_t ssrc, |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 1990 | bool mute, |
| 1991 | const VideoOptions* options, |
nisse | acd935b | 2016-11-11 03:55:13 -0800 | [diff] [blame] | 1992 | rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 1993 | return InvokeOnWorker(RTC_FROM_HERE, |
| 1994 | Bind(&VideoMediaChannel::SetVideoSend, media_channel(), |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 1995 | ssrc, mute, options, source)); |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1996 | } |
| 1997 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1998 | webrtc::RtpParameters VideoChannel::GetRtpSendParameters(uint32_t ssrc) const { |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 1999 | return worker_thread()->Invoke<webrtc::RtpParameters>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2000 | RTC_FROM_HERE, Bind(&VideoChannel::GetRtpSendParameters_w, this, ssrc)); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 2001 | } |
| 2002 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 2003 | webrtc::RtpParameters VideoChannel::GetRtpSendParameters_w( |
| 2004 | uint32_t ssrc) const { |
| 2005 | return media_channel()->GetRtpSendParameters(ssrc); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 2006 | } |
| 2007 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 2008 | bool VideoChannel::SetRtpSendParameters( |
| 2009 | uint32_t ssrc, |
| 2010 | const webrtc::RtpParameters& parameters) { |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 2011 | return InvokeOnWorker( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2012 | RTC_FROM_HERE, |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 2013 | Bind(&VideoChannel::SetRtpSendParameters_w, this, ssrc, parameters)); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 2014 | } |
| 2015 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 2016 | bool VideoChannel::SetRtpSendParameters_w(uint32_t ssrc, |
| 2017 | webrtc::RtpParameters parameters) { |
| 2018 | return media_channel()->SetRtpSendParameters(ssrc, parameters); |
| 2019 | } |
| 2020 | |
| 2021 | webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters( |
| 2022 | uint32_t ssrc) const { |
| 2023 | return worker_thread()->Invoke<webrtc::RtpParameters>( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2024 | RTC_FROM_HERE, |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 2025 | Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc)); |
| 2026 | } |
| 2027 | |
| 2028 | webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w( |
| 2029 | uint32_t ssrc) const { |
| 2030 | return media_channel()->GetRtpReceiveParameters(ssrc); |
| 2031 | } |
| 2032 | |
| 2033 | bool VideoChannel::SetRtpReceiveParameters( |
| 2034 | uint32_t ssrc, |
| 2035 | const webrtc::RtpParameters& parameters) { |
| 2036 | return InvokeOnWorker( |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2037 | RTC_FROM_HERE, |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 2038 | Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters)); |
| 2039 | } |
| 2040 | |
| 2041 | bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc, |
| 2042 | webrtc::RtpParameters parameters) { |
| 2043 | return media_channel()->SetRtpReceiveParameters(ssrc, parameters); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 2044 | } |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 2045 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2046 | void VideoChannel::UpdateMediaSendRecvState_w() { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2047 | // Send outgoing data if we're the active call, we have the remote content, |
| 2048 | // and we have had some form of connectivity. |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2049 | bool send = IsReadyToSendMedia_w(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2050 | if (!media_channel()->SetSend(send)) { |
| 2051 | LOG(LS_ERROR) << "Failed to SetSend on video channel"; |
| 2052 | // TODO(gangji): Report error back to server. |
| 2053 | } |
| 2054 | |
Peter Boström | 34fbfff | 2015-09-24 19:20:30 +0200 | [diff] [blame] | 2055 | LOG(LS_INFO) << "Changing video state, send=" << send; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2056 | } |
| 2057 | |
pbos@webrtc.org | 058b1f1 | 2015-03-04 08:54:32 +0000 | [diff] [blame] | 2058 | bool VideoChannel::GetStats(VideoMediaInfo* stats) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2059 | return InvokeOnWorker(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats, |
| 2060 | media_channel(), stats)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2061 | } |
| 2062 | |
| 2063 | void VideoChannel::StartMediaMonitor(int cms) { |
| 2064 | media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(), |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 2065 | rtc::Thread::Current())); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2066 | media_monitor_->SignalUpdate.connect( |
| 2067 | this, &VideoChannel::OnMediaMonitorUpdate); |
| 2068 | media_monitor_->Start(cms); |
| 2069 | } |
| 2070 | |
| 2071 | void VideoChannel::StopMediaMonitor() { |
| 2072 | if (media_monitor_) { |
| 2073 | media_monitor_->Stop(); |
| 2074 | media_monitor_.reset(); |
| 2075 | } |
| 2076 | } |
| 2077 | |
| 2078 | const ContentInfo* VideoChannel::GetFirstContent( |
| 2079 | const SessionDescription* sdesc) { |
| 2080 | return GetFirstVideoContent(sdesc); |
| 2081 | } |
| 2082 | |
| 2083 | bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2084 | ContentAction action, |
| 2085 | std::string* error_desc) { |
Peter Boström | 9f45a45 | 2015-12-08 13:25:57 +0100 | [diff] [blame] | 2086 | TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2087 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2088 | LOG(LS_INFO) << "Setting local video description"; |
| 2089 | |
| 2090 | const VideoContentDescription* video = |
| 2091 | static_cast<const VideoContentDescription*>(content); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2092 | RTC_DCHECK(video != NULL); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2093 | if (!video) { |
| 2094 | SafeSetError("Can't find video content in local description.", error_desc); |
| 2095 | return false; |
| 2096 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2097 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 2098 | if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2099 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2100 | } |
| 2101 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2102 | VideoRecvParameters recv_params = last_recv_params_; |
| 2103 | RtpParametersFromMediaDescription(video, &recv_params); |
| 2104 | if (!media_channel()->SetRecvParameters(recv_params)) { |
| 2105 | SafeSetError("Failed to set local video description recv parameters.", |
| 2106 | error_desc); |
| 2107 | return false; |
| 2108 | } |
| 2109 | for (const VideoCodec& codec : video->codecs()) { |
| 2110 | bundle_filter()->AddPayloadType(codec.id); |
| 2111 | } |
| 2112 | last_recv_params_ = recv_params; |
| 2113 | |
| 2114 | // TODO(pthatcher): Move local streams into VideoSendParameters, and |
| 2115 | // only give it to the media channel once we have a remote |
| 2116 | // description too (without a remote description, we won't be able |
| 2117 | // to send them anyway). |
| 2118 | if (!UpdateLocalStreams_w(video->streams(), action, error_desc)) { |
| 2119 | SafeSetError("Failed to set local video description streams.", error_desc); |
| 2120 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2121 | } |
| 2122 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2123 | set_local_content_direction(content->direction()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2124 | UpdateMediaSendRecvState_w(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2125 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2126 | } |
| 2127 | |
| 2128 | bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2129 | ContentAction action, |
| 2130 | std::string* error_desc) { |
Peter Boström | 9f45a45 | 2015-12-08 13:25:57 +0100 | [diff] [blame] | 2131 | TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2132 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2133 | LOG(LS_INFO) << "Setting remote video description"; |
| 2134 | |
| 2135 | const VideoContentDescription* video = |
| 2136 | static_cast<const VideoContentDescription*>(content); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2137 | RTC_DCHECK(video != NULL); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2138 | if (!video) { |
| 2139 | SafeSetError("Can't find video content in remote description.", error_desc); |
| 2140 | return false; |
| 2141 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2142 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 2143 | if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2144 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2145 | } |
| 2146 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2147 | VideoSendParameters send_params = last_send_params_; |
| 2148 | RtpSendParametersFromMediaDescription(video, &send_params); |
| 2149 | if (video->conference_mode()) { |
nisse | 4b4dc86 | 2016-02-17 05:25:36 -0800 | [diff] [blame] | 2150 | send_params.conference_mode = true; |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2151 | } |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 2152 | |
| 2153 | bool parameters_applied = media_channel()->SetSendParameters(send_params); |
| 2154 | |
| 2155 | if (!parameters_applied) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2156 | SafeSetError("Failed to set remote video description send parameters.", |
| 2157 | error_desc); |
| 2158 | return false; |
| 2159 | } |
| 2160 | last_send_params_ = send_params; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2161 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2162 | // TODO(pthatcher): Move remote streams into VideoRecvParameters, |
| 2163 | // and only give it to the media channel once we have a local |
| 2164 | // description too (without a local description, we won't be able to |
| 2165 | // recv them anyway). |
| 2166 | if (!UpdateRemoteStreams_w(video->streams(), action, error_desc)) { |
| 2167 | SafeSetError("Failed to set remote video description streams.", error_desc); |
| 2168 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2169 | } |
| 2170 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2171 | if (video->rtp_header_extensions_set()) { |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 2172 | MaybeCacheRtpAbsSendTimeHeaderExtension_w(video->rtp_header_extensions()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2173 | } |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2174 | |
| 2175 | set_remote_content_direction(content->direction()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2176 | UpdateMediaSendRecvState_w(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2177 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2178 | } |
| 2179 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 2180 | void VideoChannel::OnMessage(rtc::Message *pmsg) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2181 | switch (pmsg->message_id) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2182 | case MSG_CHANNEL_ERROR: { |
| 2183 | const VideoChannelErrorMessageData* data = |
| 2184 | static_cast<VideoChannelErrorMessageData*>(pmsg->pdata); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2185 | delete data; |
| 2186 | break; |
| 2187 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2188 | default: |
| 2189 | BaseChannel::OnMessage(pmsg); |
| 2190 | break; |
| 2191 | } |
| 2192 | } |
| 2193 | |
| 2194 | void VideoChannel::OnConnectionMonitorUpdate( |
pthatcher@webrtc.org | b4aac13 | 2015-03-13 18:25:21 +0000 | [diff] [blame] | 2195 | ConnectionMonitor* monitor, const std::vector<ConnectionInfo> &infos) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2196 | SignalConnectionMonitor(this, infos); |
| 2197 | } |
| 2198 | |
| 2199 | // TODO(pthatcher): Look into removing duplicate code between |
| 2200 | // audio, video, and data, perhaps by using templates. |
| 2201 | void VideoChannel::OnMediaMonitorUpdate( |
| 2202 | VideoMediaChannel* media_channel, const VideoMediaInfo &info) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2203 | RTC_DCHECK(media_channel == this->media_channel()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2204 | SignalMediaMonitor(this, info); |
| 2205 | } |
| 2206 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 2207 | void VideoChannel::GetSrtpCryptoSuites_n( |
| 2208 | std::vector<int>* crypto_suites) const { |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 2209 | GetSupportedVideoCryptoSuites(crypto_options(), crypto_suites); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2210 | } |
| 2211 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2212 | RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread, |
| 2213 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 2214 | rtc::Thread* signaling_thread, |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2215 | DataMediaChannel* media_channel, |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2216 | const std::string& content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 2217 | bool rtcp_mux_required, |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2218 | bool srtp_required) |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 2219 | : BaseChannel(worker_thread, |
| 2220 | network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 2221 | signaling_thread, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 2222 | media_channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 2223 | content_name, |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 2224 | rtcp_mux_required, |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2225 | srtp_required) {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2226 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2227 | RtpDataChannel::~RtpDataChannel() { |
| 2228 | TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2229 | StopMediaMonitor(); |
| 2230 | // this can't be done in the base class, since it calls a virtual |
| 2231 | DisableMedia_w(); |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 2232 | |
| 2233 | Deinit(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2234 | } |
| 2235 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 2236 | bool RtpDataChannel::Init_w( |
| 2237 | DtlsTransportInternal* rtp_dtls_transport, |
| 2238 | DtlsTransportInternal* rtcp_dtls_transport, |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 2239 | rtc::PacketTransportInternal* rtp_packet_transport, |
| 2240 | rtc::PacketTransportInternal* rtcp_packet_transport) { |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 2241 | if (!BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport, |
| 2242 | rtp_packet_transport, rtcp_packet_transport)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2243 | return false; |
| 2244 | } |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2245 | media_channel()->SignalDataReceived.connect(this, |
| 2246 | &RtpDataChannel::OnDataReceived); |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 2247 | media_channel()->SignalReadyToSend.connect( |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2248 | this, &RtpDataChannel::OnDataChannelReadyToSend); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2249 | return true; |
| 2250 | } |
| 2251 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2252 | bool RtpDataChannel::SendData(const SendDataParams& params, |
| 2253 | const rtc::CopyOnWriteBuffer& payload, |
| 2254 | SendDataResult* result) { |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2255 | return InvokeOnWorker( |
| 2256 | RTC_FROM_HERE, Bind(&DataMediaChannel::SendData, media_channel(), params, |
| 2257 | payload, result)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2258 | } |
| 2259 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2260 | const ContentInfo* RtpDataChannel::GetFirstContent( |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2261 | const SessionDescription* sdesc) { |
| 2262 | return GetFirstDataContent(sdesc); |
| 2263 | } |
| 2264 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2265 | bool RtpDataChannel::CheckDataChannelTypeFromContent( |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2266 | const DataContentDescription* content, |
| 2267 | std::string* error_desc) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2268 | bool is_sctp = ((content->protocol() == kMediaProtocolSctp) || |
| 2269 | (content->protocol() == kMediaProtocolDtlsSctp)); |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2270 | // It's been set before, but doesn't match. That's bad. |
| 2271 | if (is_sctp) { |
| 2272 | SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.", |
| 2273 | error_desc); |
| 2274 | return false; |
| 2275 | } |
| 2276 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2277 | } |
| 2278 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2279 | bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content, |
| 2280 | ContentAction action, |
| 2281 | std::string* error_desc) { |
| 2282 | TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2283 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2284 | LOG(LS_INFO) << "Setting local data description"; |
| 2285 | |
| 2286 | const DataContentDescription* data = |
| 2287 | static_cast<const DataContentDescription*>(content); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2288 | RTC_DCHECK(data != NULL); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2289 | if (!data) { |
| 2290 | SafeSetError("Can't find data content in local description.", error_desc); |
| 2291 | return false; |
| 2292 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2293 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2294 | if (!CheckDataChannelTypeFromContent(data, error_desc)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2295 | return false; |
| 2296 | } |
| 2297 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2298 | if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) { |
| 2299 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2300 | } |
| 2301 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2302 | DataRecvParameters recv_params = last_recv_params_; |
| 2303 | RtpParametersFromMediaDescription(data, &recv_params); |
| 2304 | if (!media_channel()->SetRecvParameters(recv_params)) { |
| 2305 | SafeSetError("Failed to set remote data description recv parameters.", |
| 2306 | error_desc); |
| 2307 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2308 | } |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2309 | for (const DataCodec& codec : data->codecs()) { |
| 2310 | bundle_filter()->AddPayloadType(codec.id); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2311 | } |
| 2312 | last_recv_params_ = recv_params; |
| 2313 | |
| 2314 | // TODO(pthatcher): Move local streams into DataSendParameters, and |
| 2315 | // only give it to the media channel once we have a remote |
| 2316 | // description too (without a remote description, we won't be able |
| 2317 | // to send them anyway). |
| 2318 | if (!UpdateLocalStreams_w(data->streams(), action, error_desc)) { |
| 2319 | SafeSetError("Failed to set local data description streams.", error_desc); |
| 2320 | return false; |
| 2321 | } |
| 2322 | |
| 2323 | set_local_content_direction(content->direction()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2324 | UpdateMediaSendRecvState_w(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2325 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2326 | } |
| 2327 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2328 | bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| 2329 | ContentAction action, |
| 2330 | std::string* error_desc) { |
| 2331 | TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w"); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2332 | RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2333 | |
| 2334 | const DataContentDescription* data = |
| 2335 | static_cast<const DataContentDescription*>(content); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2336 | RTC_DCHECK(data != NULL); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 2337 | if (!data) { |
| 2338 | SafeSetError("Can't find data content in remote description.", error_desc); |
| 2339 | return false; |
| 2340 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2341 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2342 | // If the remote data doesn't have codecs and isn't an update, it |
| 2343 | // must be empty, so ignore it. |
| 2344 | if (!data->has_codecs() && action != CA_UPDATE) { |
| 2345 | return true; |
| 2346 | } |
| 2347 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2348 | if (!CheckDataChannelTypeFromContent(data, error_desc)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2349 | return false; |
| 2350 | } |
| 2351 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2352 | LOG(LS_INFO) << "Setting remote data description"; |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2353 | if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) { |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2354 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2355 | } |
| 2356 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2357 | DataSendParameters send_params = last_send_params_; |
| 2358 | RtpSendParametersFromMediaDescription<DataCodec>(data, &send_params); |
| 2359 | if (!media_channel()->SetSendParameters(send_params)) { |
| 2360 | SafeSetError("Failed to set remote data description send parameters.", |
| 2361 | error_desc); |
| 2362 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2363 | } |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2364 | last_send_params_ = send_params; |
| 2365 | |
| 2366 | // TODO(pthatcher): Move remote streams into DataRecvParameters, |
| 2367 | // and only give it to the media channel once we have a local |
| 2368 | // description too (without a local description, we won't be able to |
| 2369 | // recv them anyway). |
| 2370 | if (!UpdateRemoteStreams_w(data->streams(), action, error_desc)) { |
| 2371 | SafeSetError("Failed to set remote data description streams.", |
| 2372 | error_desc); |
| 2373 | return false; |
| 2374 | } |
| 2375 | |
| 2376 | set_remote_content_direction(content->direction()); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2377 | UpdateMediaSendRecvState_w(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 2378 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2379 | } |
| 2380 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2381 | void RtpDataChannel::UpdateMediaSendRecvState_w() { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2382 | // Render incoming data if we're the active call, and we have the local |
| 2383 | // content. We receive data on the default channel and multiplexed streams. |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2384 | bool recv = IsReadyToReceiveMedia_w(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2385 | if (!media_channel()->SetReceive(recv)) { |
| 2386 | LOG(LS_ERROR) << "Failed to SetReceive on data channel"; |
| 2387 | } |
| 2388 | |
| 2389 | // Send outgoing data if we're the active call, we have the remote content, |
| 2390 | // and we have had some form of connectivity. |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2391 | bool send = IsReadyToSendMedia_w(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2392 | if (!media_channel()->SetSend(send)) { |
| 2393 | LOG(LS_ERROR) << "Failed to SetSend on data channel"; |
| 2394 | } |
| 2395 | |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 2396 | // Trigger SignalReadyToSendData asynchronously. |
| 2397 | OnDataChannelReadyToSend(send); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2398 | |
| 2399 | LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send; |
| 2400 | } |
| 2401 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2402 | void RtpDataChannel::OnMessage(rtc::Message* pmsg) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2403 | switch (pmsg->message_id) { |
| 2404 | case MSG_READYTOSENDDATA: { |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 2405 | DataChannelReadyToSendMessageData* data = |
| 2406 | static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata); |
wu@webrtc.org | 07a6fbe | 2013-11-04 18:41:34 +0000 | [diff] [blame] | 2407 | ready_to_send_data_ = data->data(); |
| 2408 | SignalReadyToSendData(ready_to_send_data_); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2409 | delete data; |
| 2410 | break; |
| 2411 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2412 | case MSG_DATARECEIVED: { |
| 2413 | DataReceivedMessageData* data = |
| 2414 | static_cast<DataReceivedMessageData*>(pmsg->pdata); |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2415 | SignalDataReceived(data->params, data->payload); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2416 | delete data; |
| 2417 | break; |
| 2418 | } |
| 2419 | case MSG_CHANNEL_ERROR: { |
| 2420 | const DataChannelErrorMessageData* data = |
| 2421 | static_cast<DataChannelErrorMessageData*>(pmsg->pdata); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2422 | delete data; |
| 2423 | break; |
| 2424 | } |
| 2425 | default: |
| 2426 | BaseChannel::OnMessage(pmsg); |
| 2427 | break; |
| 2428 | } |
| 2429 | } |
| 2430 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2431 | void RtpDataChannel::OnConnectionMonitorUpdate( |
| 2432 | ConnectionMonitor* monitor, |
| 2433 | const std::vector<ConnectionInfo>& infos) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2434 | SignalConnectionMonitor(this, infos); |
| 2435 | } |
| 2436 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2437 | void RtpDataChannel::StartMediaMonitor(int cms) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2438 | media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(), |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 2439 | rtc::Thread::Current())); |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2440 | media_monitor_->SignalUpdate.connect(this, |
| 2441 | &RtpDataChannel::OnMediaMonitorUpdate); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2442 | media_monitor_->Start(cms); |
| 2443 | } |
| 2444 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2445 | void RtpDataChannel::StopMediaMonitor() { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2446 | if (media_monitor_) { |
| 2447 | media_monitor_->Stop(); |
| 2448 | media_monitor_->SignalUpdate.disconnect(this); |
| 2449 | media_monitor_.reset(); |
| 2450 | } |
| 2451 | } |
| 2452 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2453 | void RtpDataChannel::OnMediaMonitorUpdate(DataMediaChannel* media_channel, |
| 2454 | const DataMediaInfo& info) { |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 2455 | RTC_DCHECK(media_channel == this->media_channel()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2456 | SignalMediaMonitor(this, info); |
| 2457 | } |
| 2458 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2459 | void RtpDataChannel::OnDataReceived(const ReceiveDataParams& params, |
| 2460 | const char* data, |
| 2461 | size_t len) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2462 | DataReceivedMessageData* msg = new DataReceivedMessageData( |
| 2463 | params, data, len); |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2464 | signaling_thread()->Post(RTC_FROM_HERE, this, MSG_DATARECEIVED, msg); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2465 | } |
| 2466 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2467 | void RtpDataChannel::OnDataChannelError(uint32_t ssrc, |
| 2468 | DataMediaChannel::Error err) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2469 | DataChannelErrorMessageData* data = new DataChannelErrorMessageData( |
| 2470 | ssrc, err); |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2471 | signaling_thread()->Post(RTC_FROM_HERE, this, MSG_CHANNEL_ERROR, data); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2472 | } |
| 2473 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2474 | void RtpDataChannel::OnDataChannelReadyToSend(bool writable) { |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 2475 | // This is usded for congestion control to indicate that the stream is ready |
| 2476 | // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates |
| 2477 | // that the transport channel is ready. |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame] | 2478 | signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 2479 | new DataChannelReadyToSendMessageData(writable)); |
| 2480 | } |
| 2481 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 2482 | void RtpDataChannel::GetSrtpCryptoSuites_n( |
| 2483 | std::vector<int>* crypto_suites) const { |
jbauch | cb56065 | 2016-08-04 05:20:32 -0700 | [diff] [blame] | 2484 | GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2485 | } |
| 2486 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2487 | } // namespace cricket |