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 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 11 | #ifndef PC_CHANNEL_H_ |
| 12 | #define PC_CHANNEL_H_ |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 13 | |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 14 | #include <map> |
kwiberg | 3102294 | 2016-03-11 14:18:21 -0800 | [diff] [blame] | 15 | #include <memory> |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 16 | #include <set> |
kjellander | a96e2d7 | 2016-02-04 23:52:28 -0800 | [diff] [blame] | 17 | #include <string> |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 18 | #include <utility> |
kjellander | a96e2d7 | 2016-02-04 23:52:28 -0800 | [diff] [blame] | 19 | #include <vector> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 20 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 21 | #include "api/call/audio_sink.h" |
Danil Chapovalov | 8931345 | 2019-11-29 12:56:43 +0100 | [diff] [blame] | 22 | #include "api/function_view.h" |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 23 | #include "api/jsep.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 24 | #include "api/rtp_receiver_interface.h" |
Niels Möller | 65f17ca | 2019-09-12 13:59:36 +0200 | [diff] [blame] | 25 | #include "api/transport/media/media_transport_config.h" |
Niels Möller | c6ce9c5 | 2018-05-11 11:15:30 +0200 | [diff] [blame] | 26 | #include "api/video/video_sink_interface.h" |
Niels Möller | 0327c2d | 2018-05-21 14:09:31 +0200 | [diff] [blame] | 27 | #include "api/video/video_source_interface.h" |
Zhi Huang | 365381f | 2018-04-13 16:44:34 -0700 | [diff] [blame] | 28 | #include "call/rtp_packet_sink_interface.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 29 | #include "media/base/media_channel.h" |
| 30 | #include "media/base/media_engine.h" |
| 31 | #include "media/base/stream_params.h" |
| 32 | #include "p2p/base/dtls_transport_internal.h" |
| 33 | #include "p2p/base/packet_transport_internal.h" |
| 34 | #include "pc/channel_interface.h" |
| 35 | #include "pc/dtls_srtp_transport.h" |
| 36 | #include "pc/media_session.h" |
| 37 | #include "pc/rtp_transport.h" |
| 38 | #include "pc/srtp_filter.h" |
| 39 | #include "pc/srtp_transport.h" |
| 40 | #include "rtc_base/async_invoker.h" |
| 41 | #include "rtc_base/async_udp_socket.h" |
| 42 | #include "rtc_base/critical_section.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 43 | #include "rtc_base/network.h" |
Artem Titov | e41c433 | 2018-07-25 15:04:28 +0200 | [diff] [blame] | 44 | #include "rtc_base/third_party/sigslot/sigslot.h" |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 45 | #include "rtc_base/unique_id_generator.h" |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 46 | |
| 47 | namespace webrtc { |
| 48 | class AudioSinkInterface; |
Anton Sukhanov | 98a462c | 2018-10-17 13:15:42 -0700 | [diff] [blame] | 49 | class MediaTransportInterface; |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 50 | } // namespace webrtc |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 51 | |
| 52 | namespace cricket { |
| 53 | |
| 54 | struct CryptoParams; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 55 | |
deadbeef | 062ce9f | 2016-08-26 21:42:15 -0700 | [diff] [blame] | 56 | // BaseChannel contains logic common to voice and video, including enable, |
| 57 | // marshaling calls to a worker and network threads, and connection and media |
| 58 | // monitors. |
| 59 | // |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 60 | // BaseChannel assumes signaling and other threads are allowed to make |
| 61 | // synchronous calls to the worker thread, the worker thread makes synchronous |
| 62 | // calls only to the network thread, and the network thread can't be blocked by |
| 63 | // other threads. |
| 64 | // All methods with _n suffix must be called on network thread, |
deadbeef | 062ce9f | 2016-08-26 21:42:15 -0700 | [diff] [blame] | 65 | // methods with _w suffix on worker thread |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 66 | // and methods with _s suffix on signaling thread. |
| 67 | // Network and worker threads may be the same thread. |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 68 | // |
| 69 | // WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS! |
| 70 | // This is required to avoid a data race between the destructor modifying the |
| 71 | // vtable, and the media channel's thread using BaseChannel as the |
| 72 | // NetworkInterface. |
| 73 | |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 74 | class BaseChannel : public ChannelInterface, |
| 75 | public rtc::MessageHandler, |
Zhi Huang | 365381f | 2018-04-13 16:44:34 -0700 | [diff] [blame] | 76 | public sigslot::has_slots<>, |
| 77 | public MediaChannel::NetworkInterface, |
Bjorn A Mellem | 7a9a092 | 2019-11-26 09:19:40 -0800 | [diff] [blame] | 78 | public webrtc::RtpPacketSinkInterface { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 79 | public: |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 80 | // If |srtp_required| is true, the channel will not send or receive any |
| 81 | // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP). |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 82 | // The BaseChannel does not own the UniqueRandomIdGenerator so it is the |
| 83 | // responsibility of the user to ensure it outlives this object. |
Zhi Huang | e830e68 | 2018-03-30 10:48:35 -0700 | [diff] [blame] | 84 | // TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists |
| 85 | // which will make it easier to change the constructor. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 86 | BaseChannel(rtc::Thread* worker_thread, |
| 87 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 88 | rtc::Thread* signaling_thread, |
Steve Anton | 8699a32 | 2017-11-06 15:53:33 -0800 | [diff] [blame] | 89 | std::unique_ptr<MediaChannel> media_channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 90 | const std::string& content_name, |
Zhi Huang | e830e68 | 2018-03-30 10:48:35 -0700 | [diff] [blame] | 91 | bool srtp_required, |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 92 | webrtc::CryptoOptions crypto_options, |
| 93 | rtc::UniqueRandomIdGenerator* ssrc_generator); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 94 | virtual ~BaseChannel(); |
Anton Sukhanov | 4f08faa | 2019-05-21 11:12:57 -0700 | [diff] [blame] | 95 | virtual void Init_w( |
| 96 | webrtc::RtpTransportInternal* rtp_transport, |
| 97 | const webrtc::MediaTransportConfig& media_transport_config); |
Zhi Huang | 2dfc42d | 2017-12-04 13:38:48 -0800 | [diff] [blame] | 98 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 99 | // Deinit may be called multiple times and is simply ignored if it's already |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 100 | // done. |
| 101 | void Deinit(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 102 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 103 | rtc::Thread* worker_thread() const { return worker_thread_; } |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 104 | rtc::Thread* network_thread() const { return network_thread_; } |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 105 | const std::string& content_name() const override { return content_name_; } |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 106 | // TODO(deadbeef): This is redundant; remove this. |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 107 | const std::string& transport_name() const override { return transport_name_; } |
| 108 | bool enabled() const override { return enabled_; } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 109 | |
Zhi Huang | cf990f5 | 2017-09-22 12:12:30 -0700 | [diff] [blame] | 110 | // This function returns true if using SRTP (DTLS-based keying or SDES). |
Zhi Huang | e830e68 | 2018-03-30 10:48:35 -0700 | [diff] [blame] | 111 | bool srtp_active() const { |
| 112 | return rtp_transport_ && rtp_transport_->IsSrtpActive(); |
| 113 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 114 | |
| 115 | bool writable() const { return writable_; } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 116 | |
Zhi Huang | 2dfc42d | 2017-12-04 13:38:48 -0800 | [diff] [blame] | 117 | // Set an RTP level transport which could be an RtpTransport without |
| 118 | // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP. |
| 119 | // This can be called from any thread and it hops to the network thread |
| 120 | // internally. It would replace the |SetTransports| and its variants. |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 121 | bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override; |
Zhi Huang | 2dfc42d | 2017-12-04 13:38:48 -0800 | [diff] [blame] | 122 | |
Bjorn A Mellem | 3a1b927 | 2019-05-24 16:13:08 -0700 | [diff] [blame] | 123 | webrtc::RtpTransportInternal* rtp_transport() const { return rtp_transport_; } |
| 124 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 125 | // Channel control |
| 126 | bool SetLocalContent(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 127 | webrtc::SdpType type, |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 128 | std::string* error_desc) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 129 | bool SetRemoteContent(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 130 | webrtc::SdpType type, |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 131 | std::string* error_desc) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 132 | |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 133 | bool Enable(bool enable) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 134 | |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 135 | const std::vector<StreamParams>& local_streams() const override { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 136 | return local_streams_; |
| 137 | } |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 138 | const std::vector<StreamParams>& remote_streams() const override { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 139 | return remote_streams_; |
| 140 | } |
| 141 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 142 | sigslot::signal2<BaseChannel*, bool> SignalDtlsSrtpSetupFailure; |
| 143 | void SignalDtlsSrtpSetupFailure_n(bool rtcp); |
| 144 | void SignalDtlsSrtpSetupFailure_s(bool rtcp); |
pthatcher@webrtc.org | 4eeef58 | 2015-03-16 19:34:23 +0000 | [diff] [blame] | 145 | |
buildbot@webrtc.org | 6bfd619 | 2014-05-15 16:15:59 +0000 | [diff] [blame] | 146 | // Used for latency measurements. |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 147 | sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() override { |
| 148 | return SignalFirstPacketReceived_; |
| 149 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 150 | |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 151 | // Forward SignalSentPacket to worker thread. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 152 | sigslot::signal1<const rtc::SentPacket&> SignalSentPacket; |
| 153 | |
deadbeef | ac22f70 | 2017-01-12 21:59:29 -0800 | [diff] [blame] | 154 | // Emitted whenever rtcp-mux is fully negotiated and the rtcp-transport can |
| 155 | // be destroyed. |
| 156 | // Fired on the network thread. |
| 157 | sigslot::signal1<const std::string&> SignalRtcpMuxFullyActive; |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 158 | |
zstein | 56162b9 | 2017-04-24 16:54:35 -0700 | [diff] [blame] | 159 | // From RtpTransport - public for testing only |
| 160 | void OnTransportReadyToSend(bool ready); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 161 | |
guoweis@webrtc.org | 4f85288 | 2015-03-12 20:09:44 +0000 | [diff] [blame] | 162 | // Only public for unit tests. Otherwise, consider protected. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 163 | int SetOption(SocketType type, rtc::Socket::Option o, int val) override; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 164 | int SetOption_n(SocketType type, rtc::Socket::Option o, int val); |
guoweis@webrtc.org | 4f85288 | 2015-03-12 20:09:44 +0000 | [diff] [blame] | 165 | |
Zhi Huang | 365381f | 2018-04-13 16:44:34 -0700 | [diff] [blame] | 166 | // RtpPacketSinkInterface overrides. |
| 167 | void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override; |
zstein | 3dcf0e9 | 2017-06-01 13:22:42 -0700 | [diff] [blame] | 168 | |
Steve Anton | 593e325 | 2017-12-15 11:44:48 -0800 | [diff] [blame] | 169 | // Used by the RTCStatsCollector tests to set the transport name without |
| 170 | // creating RtpTransports. |
| 171 | void set_transport_name_for_testing(const std::string& transport_name) { |
| 172 | transport_name_ = transport_name; |
| 173 | } |
| 174 | |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 175 | MediaChannel* media_channel() const override { return media_channel_.get(); } |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 176 | |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 177 | protected: |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 178 | bool was_ever_writable() const { return was_ever_writable_; } |
Steve Anton | 4e70a72 | 2017-11-28 14:57:10 -0800 | [diff] [blame] | 179 | void set_local_content_direction(webrtc::RtpTransceiverDirection direction) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 180 | local_content_direction_ = direction; |
| 181 | } |
Steve Anton | 4e70a72 | 2017-11-28 14:57:10 -0800 | [diff] [blame] | 182 | void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 183 | remote_content_direction_ = direction; |
| 184 | } |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 185 | // These methods verify that: |
| 186 | // * The required content description directions have been set. |
| 187 | // * The channel is enabled. |
| 188 | // * And for sending: |
| 189 | // - The SRTP filter is active if it's needed. |
| 190 | // - The transport has been writable before, meaning it should be at least |
| 191 | // possible to succeed in sending a packet. |
| 192 | // |
| 193 | // When any of these properties change, UpdateMediaSendRecvState_w should be |
| 194 | // called. |
| 195 | bool IsReadyToReceiveMedia_w() const; |
| 196 | bool IsReadyToSendMedia_w() const; |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 197 | rtc::Thread* signaling_thread() { return signaling_thread_; } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 198 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 199 | void FlushRtcpMessages_n(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 200 | |
| 201 | // NetworkInterface implementation, called by MediaEngine |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 202 | bool SendPacket(rtc::CopyOnWriteBuffer* packet, |
| 203 | const rtc::PacketOptions& options) override; |
| 204 | bool SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| 205 | const rtc::PacketOptions& options) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 206 | |
Zhi Huang | cd3fc5d | 2017-11-29 10:41:57 -0800 | [diff] [blame] | 207 | // From RtpTransportInternal |
| 208 | void OnWritableState(bool writable); |
Guo-wei Shieh | 1218d7a | 2015-12-05 09:59:56 -0800 | [diff] [blame] | 209 | |
Danil Chapovalov | 66cadcc | 2018-06-19 16:47:43 +0200 | [diff] [blame] | 210 | void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route); |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 211 | |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 212 | bool PacketIsRtcp(const rtc::PacketTransportInternal* transport, |
johan | d89ab14 | 2016-10-25 10:50:32 -0700 | [diff] [blame] | 213 | const char* data, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 214 | size_t len); |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 215 | bool SendPacket(bool rtcp, |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 216 | rtc::CopyOnWriteBuffer* packet, |
stefan | c1aeaf0 | 2015-10-15 07:26:07 -0700 | [diff] [blame] | 217 | const rtc::PacketOptions& options); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 218 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 219 | void EnableMedia_w(); |
| 220 | void DisableMedia_w(); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 221 | |
| 222 | // Performs actions if the RTP/RTCP writable state changed. This should |
| 223 | // be called whenever a channel's writable state changes or when RTCP muxing |
| 224 | // becomes active/inactive. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 225 | void UpdateWritableState_n(); |
| 226 | void ChannelWritable_n(); |
| 227 | void ChannelNotWritable_n(); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 228 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 229 | bool AddRecvStream_w(const StreamParams& sp); |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 230 | bool RemoveRecvStream_w(uint32_t ssrc); |
Saurav Das | ff27da5 | 2019-09-20 11:05:30 -0700 | [diff] [blame] | 231 | void ResetUnsignaledRecvStream_w(); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 232 | bool AddSendStream_w(const StreamParams& sp); |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 233 | bool RemoveSendStream_w(uint32_t ssrc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 234 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 235 | // Should be called whenever the conditions for |
| 236 | // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied). |
| 237 | // Updates the send/recv state of the media channel. |
| 238 | void UpdateMediaSendRecvState(); |
| 239 | virtual void UpdateMediaSendRecvState_w() = 0; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 240 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 241 | bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 242 | webrtc::SdpType type, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 243 | std::string* error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 244 | bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 245 | webrtc::SdpType type, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 246 | std::string* error_desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 247 | virtual bool SetLocalContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 248 | webrtc::SdpType type, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 249 | std::string* error_desc) = 0; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 250 | virtual bool SetRemoteContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 251 | webrtc::SdpType type, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 252 | std::string* error_desc) = 0; |
jbauch | 5869f50 | 2017-06-29 12:31:36 -0700 | [diff] [blame] | 253 | // Return a list of RTP header extensions with the non-encrypted extensions |
| 254 | // removed depending on the current crypto_options_ and only if both the |
| 255 | // non-encrypted and encrypted extension is present for the same URI. |
| 256 | RtpHeaderExtensions GetFilteredRtpHeaderExtensions( |
| 257 | const RtpHeaderExtensions& extensions); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 258 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 259 | // From MessageHandler |
rlester | ec9d187 | 2015-10-27 14:22:16 -0700 | [diff] [blame] | 260 | void OnMessage(rtc::Message* pmsg) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 261 | |
stefan | f79ade1 | 2017-06-02 06:44:03 -0700 | [diff] [blame] | 262 | // Helper function template for invoking methods on the worker thread. |
Danil Chapovalov | 8931345 | 2019-11-29 12:56:43 +0100 | [diff] [blame] | 263 | template <class T> |
| 264 | T InvokeOnWorker(const rtc::Location& posted_from, |
| 265 | rtc::FunctionView<T()> functor) { |
stefan | f79ade1 | 2017-06-02 06:44:03 -0700 | [diff] [blame] | 266 | return worker_thread_->Invoke<T>(posted_from, functor); |
sergeyu@chromium.org | 9cf037b | 2014-02-07 19:03:26 +0000 | [diff] [blame] | 267 | } |
| 268 | |
zstein | 3dcf0e9 | 2017-06-01 13:22:42 -0700 | [diff] [blame] | 269 | void AddHandledPayloadType(int payload_type); |
| 270 | |
Steve Anton | be2e5f7 | 2019-09-06 16:26:02 -0700 | [diff] [blame] | 271 | void ClearHandledPayloadTypes(); |
| 272 | |
Zhi Huang | 365381f | 2018-04-13 16:44:34 -0700 | [diff] [blame] | 273 | void UpdateRtpHeaderExtensionMap( |
| 274 | const RtpHeaderExtensions& header_extensions); |
| 275 | |
| 276 | bool RegisterRtpDemuxerSink(); |
| 277 | |
Piotr (Peter) Slatala | 309aafe | 2019-01-15 14:24:34 -0800 | [diff] [blame] | 278 | bool has_received_packet_ = false; |
| 279 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 280 | private: |
Zhi Huang | 365381f | 2018-04-13 16:44:34 -0700 | [diff] [blame] | 281 | bool ConnectToRtpTransport(); |
Zhi Huang | cd3fc5d | 2017-11-29 10:41:57 -0800 | [diff] [blame] | 282 | void DisconnectFromRtpTransport(); |
Zhi Huang | cd3fc5d | 2017-11-29 10:41:57 -0800 | [diff] [blame] | 283 | void SignalSentPacket_n(const rtc::SentPacket& sent_packet); |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 284 | bool IsReadyToSendMedia_n() const; |
Piotr (Peter) Slatala | 179a392 | 2018-11-16 09:57:58 -0800 | [diff] [blame] | 285 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 286 | rtc::Thread* const worker_thread_; |
| 287 | rtc::Thread* const network_thread_; |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 288 | rtc::Thread* const signaling_thread_; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 289 | rtc::AsyncInvoker invoker_; |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 290 | sigslot::signal1<ChannelInterface*> SignalFirstPacketReceived_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 291 | |
pthatcher@webrtc.org | 990a00c | 2015-03-13 18:20:33 +0000 | [diff] [blame] | 292 | const std::string content_name_; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 293 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 294 | // Won't be set when using raw packet transports. SDP-specific thing. |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 295 | std::string transport_name_; |
zhihuang | b2cdd93 | 2017-01-19 16:54:25 -0800 | [diff] [blame] | 296 | |
Zhi Huang | cd3fc5d | 2017-11-29 10:41:57 -0800 | [diff] [blame] | 297 | webrtc::RtpTransportInternal* rtp_transport_ = nullptr; |
Zhi Huang | cd3fc5d | 2017-11-29 10:41:57 -0800 | [diff] [blame] | 298 | |
Anton Sukhanov | 4f08faa | 2019-05-21 11:12:57 -0700 | [diff] [blame] | 299 | // Optional media transport configuration (experimental). |
| 300 | webrtc::MediaTransportConfig media_transport_config_; |
Anton Sukhanov | 98a462c | 2018-10-17 13:15:42 -0700 | [diff] [blame] | 301 | |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 302 | std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 303 | std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; |
deadbeef | 23d947d | 2016-08-22 16:00:30 -0700 | [diff] [blame] | 304 | bool writable_ = false; |
| 305 | bool was_ever_writable_ = false; |
deadbeef | 7af91dd | 2016-12-13 11:29:11 -0800 | [diff] [blame] | 306 | const bool srtp_required_ = true; |
Benjamin Wright | a54daf1 | 2018-10-11 15:33:17 -0700 | [diff] [blame] | 307 | webrtc::CryptoOptions crypto_options_; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 308 | |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 309 | // MediaChannel related members that should be accessed from the worker |
| 310 | // thread. |
Steve Anton | 8699a32 | 2017-11-06 15:53:33 -0800 | [diff] [blame] | 311 | std::unique_ptr<MediaChannel> media_channel_; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 312 | // Currently the |enabled_| flag is accessed from the signaling thread as |
| 313 | // well, but it can be changed only when signaling thread does a synchronous |
| 314 | // call to the worker thread, so it should be safe. |
deadbeef | 23d947d | 2016-08-22 16:00:30 -0700 | [diff] [blame] | 315 | bool enabled_ = false; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 316 | std::vector<StreamParams> local_streams_; |
| 317 | std::vector<StreamParams> remote_streams_; |
Steve Anton | 4e70a72 | 2017-11-28 14:57:10 -0800 | [diff] [blame] | 318 | webrtc::RtpTransceiverDirection local_content_direction_ = |
| 319 | webrtc::RtpTransceiverDirection::kInactive; |
| 320 | webrtc::RtpTransceiverDirection remote_content_direction_ = |
| 321 | webrtc::RtpTransceiverDirection::kInactive; |
Zhi Huang | c99b6c7 | 2017-11-10 16:44:46 -0800 | [diff] [blame] | 322 | |
Zhi Huang | 365381f | 2018-04-13 16:44:34 -0700 | [diff] [blame] | 323 | webrtc::RtpDemuxerCriteria demuxer_criteria_; |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 324 | // This generator is used to generate SSRCs for local streams. |
| 325 | // This is needed in cases where SSRCs are not negotiated or set explicitly |
| 326 | // like in Simulcast. |
| 327 | // This object is not owned by the channel so it must outlive it. |
| 328 | rtc::UniqueRandomIdGenerator* const ssrc_generator_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 329 | }; |
| 330 | |
| 331 | // VoiceChannel is a specialization that adds support for early media, DTMF, |
| 332 | // and input/output level monitoring. |
Bjorn A Mellem | 7a9a092 | 2019-11-26 09:19:40 -0800 | [diff] [blame] | 333 | class VoiceChannel : public BaseChannel { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 334 | public: |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 335 | VoiceChannel(rtc::Thread* worker_thread, |
| 336 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 337 | rtc::Thread* signaling_thread, |
Steve Anton | 8699a32 | 2017-11-06 15:53:33 -0800 | [diff] [blame] | 338 | std::unique_ptr<VoiceMediaChannel> channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 339 | const std::string& content_name, |
Zhi Huang | e830e68 | 2018-03-30 10:48:35 -0700 | [diff] [blame] | 340 | bool srtp_required, |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 341 | webrtc::CryptoOptions crypto_options, |
| 342 | rtc::UniqueRandomIdGenerator* ssrc_generator); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 343 | ~VoiceChannel(); |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 344 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 345 | // downcasts a MediaChannel |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 346 | VoiceMediaChannel* media_channel() const override { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 347 | return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel()); |
| 348 | } |
| 349 | |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 350 | cricket::MediaType media_type() const override { |
| 351 | return cricket::MEDIA_TYPE_AUDIO; |
| 352 | } |
Anton Sukhanov | 4f08faa | 2019-05-21 11:12:57 -0700 | [diff] [blame] | 353 | void Init_w( |
| 354 | webrtc::RtpTransportInternal* rtp_transport, |
| 355 | const webrtc::MediaTransportConfig& media_transport_config) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 356 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 357 | private: |
| 358 | // overrides from BaseChannel |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 359 | void UpdateMediaSendRecvState_w() override; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 360 | bool SetLocalContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 361 | webrtc::SdpType type, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 362 | std::string* error_desc) override; |
| 363 | bool SetRemoteContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 364 | webrtc::SdpType type, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 365 | std::string* error_desc) override; |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 366 | |
| 367 | // Last AudioSendParameters sent down to the media_channel() via |
| 368 | // SetSendParameters. |
| 369 | AudioSendParameters last_send_params_; |
| 370 | // Last AudioRecvParameters sent down to the media_channel() via |
| 371 | // SetRecvParameters. |
| 372 | AudioRecvParameters last_recv_params_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 373 | }; |
| 374 | |
| 375 | // VideoChannel is a specialization for video. |
| 376 | class VideoChannel : public BaseChannel { |
| 377 | public: |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 378 | VideoChannel(rtc::Thread* worker_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 379 | rtc::Thread* network_thread, |
| 380 | rtc::Thread* signaling_thread, |
Steve Anton | 8699a32 | 2017-11-06 15:53:33 -0800 | [diff] [blame] | 381 | std::unique_ptr<VideoMediaChannel> media_channel, |
deadbeef | cbecd35 | 2015-09-23 11:50:27 -0700 | [diff] [blame] | 382 | const std::string& content_name, |
Zhi Huang | e830e68 | 2018-03-30 10:48:35 -0700 | [diff] [blame] | 383 | bool srtp_required, |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 384 | webrtc::CryptoOptions crypto_options, |
| 385 | rtc::UniqueRandomIdGenerator* ssrc_generator); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 386 | ~VideoChannel(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 387 | |
Fredrik Solenberg | 4b60c73 | 2015-05-07 14:07:48 +0200 | [diff] [blame] | 388 | // downcasts a MediaChannel |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 389 | VideoMediaChannel* media_channel() const override { |
Fredrik Solenberg | 4b60c73 | 2015-05-07 14:07:48 +0200 | [diff] [blame] | 390 | return static_cast<VideoMediaChannel*>(BaseChannel::media_channel()); |
| 391 | } |
| 392 | |
stefan | f79ade1 | 2017-06-02 06:44:03 -0700 | [diff] [blame] | 393 | void FillBitrateInfo(BandwidthEstimationInfo* bwe_info); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 394 | |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 395 | cricket::MediaType media_type() const override { |
| 396 | return cricket::MEDIA_TYPE_VIDEO; |
| 397 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 398 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 399 | private: |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 400 | // overrides from BaseChannel |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 401 | void UpdateMediaSendRecvState_w() override; |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 402 | bool SetLocalContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 403 | webrtc::SdpType type, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 404 | std::string* error_desc) override; |
| 405 | bool SetRemoteContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 406 | webrtc::SdpType type, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 407 | std::string* error_desc) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 408 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 409 | // Last VideoSendParameters sent down to the media_channel() via |
| 410 | // SetSendParameters. |
| 411 | VideoSendParameters last_send_params_; |
| 412 | // Last VideoRecvParameters sent down to the media_channel() via |
| 413 | // SetRecvParameters. |
| 414 | VideoRecvParameters last_recv_params_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 415 | }; |
| 416 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 417 | // RtpDataChannel is a specialization for data. |
| 418 | class RtpDataChannel : public BaseChannel { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 419 | public: |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 420 | RtpDataChannel(rtc::Thread* worker_thread, |
| 421 | rtc::Thread* network_thread, |
zhihuang | f5b251b | 2017-01-12 19:37:48 -0800 | [diff] [blame] | 422 | rtc::Thread* signaling_thread, |
Steve Anton | 8699a32 | 2017-11-06 15:53:33 -0800 | [diff] [blame] | 423 | std::unique_ptr<DataMediaChannel> channel, |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 424 | const std::string& content_name, |
Zhi Huang | e830e68 | 2018-03-30 10:48:35 -0700 | [diff] [blame] | 425 | bool srtp_required, |
Amit Hilbuch | bcd39d4 | 2019-01-25 17:13:56 -0800 | [diff] [blame] | 426 | webrtc::CryptoOptions crypto_options, |
| 427 | rtc::UniqueRandomIdGenerator* ssrc_generator); |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 428 | ~RtpDataChannel(); |
Zhi Huang | 2dfc42d | 2017-12-04 13:38:48 -0800 | [diff] [blame] | 429 | // TODO(zhihuang): Remove this once the RtpTransport can be shared between |
| 430 | // BaseChannels. |
Steve Anton | 8699a32 | 2017-11-06 15:53:33 -0800 | [diff] [blame] | 431 | void Init_w(DtlsTransportInternal* rtp_dtls_transport, |
deadbeef | f534659 | 2017-01-24 21:51:21 -0800 | [diff] [blame] | 432 | DtlsTransportInternal* rtcp_dtls_transport, |
deadbeef | 5bd5ca3 | 2017-02-10 11:31:50 -0800 | [diff] [blame] | 433 | rtc::PacketTransportInternal* rtp_packet_transport, |
| 434 | rtc::PacketTransportInternal* rtcp_packet_transport); |
Piotr (Peter) Slatala | 309aafe | 2019-01-15 14:24:34 -0800 | [diff] [blame] | 435 | void Init_w( |
| 436 | webrtc::RtpTransportInternal* rtp_transport, |
Anton Sukhanov | 4f08faa | 2019-05-21 11:12:57 -0700 | [diff] [blame] | 437 | const webrtc::MediaTransportConfig& media_transport_config) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 438 | |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 439 | virtual bool SendData(const SendDataParams& params, |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 440 | const rtc::CopyOnWriteBuffer& payload, |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 441 | SendDataResult* result); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 442 | |
wu@webrtc.org | 07a6fbe | 2013-11-04 18:41:34 +0000 | [diff] [blame] | 443 | // Should be called on the signaling thread only. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 444 | bool ready_to_send_data() const { return ready_to_send_data_; } |
wu@webrtc.org | 07a6fbe | 2013-11-04 18:41:34 +0000 | [diff] [blame] | 445 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 446 | sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&> |
| 447 | SignalDataReceived; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 448 | // Signal for notifying when the channel becomes ready to send data. |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 449 | // That occurs when the channel is enabled, the transport is writable, |
| 450 | // both local and remote descriptions are set, and the channel is unblocked. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 451 | sigslot::signal1<bool> SignalReadyToSendData; |
Amit Hilbuch | dd9390c | 2018-11-13 16:26:05 -0800 | [diff] [blame] | 452 | cricket::MediaType media_type() const override { |
| 453 | return cricket::MEDIA_TYPE_DATA; |
| 454 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 455 | |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 456 | protected: |
| 457 | // downcasts a MediaChannel. |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 458 | DataMediaChannel* media_channel() const override { |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 459 | return static_cast<DataMediaChannel*>(BaseChannel::media_channel()); |
| 460 | } |
| 461 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 462 | private: |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 463 | struct SendDataMessageData : public rtc::MessageData { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 464 | SendDataMessageData(const SendDataParams& params, |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 465 | const rtc::CopyOnWriteBuffer* payload, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 466 | SendDataResult* result) |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 467 | : params(params), payload(payload), result(result), succeeded(false) {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 468 | |
| 469 | const SendDataParams& params; |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 470 | const rtc::CopyOnWriteBuffer* payload; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 471 | SendDataResult* result; |
| 472 | bool succeeded; |
| 473 | }; |
| 474 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 475 | struct DataReceivedMessageData : public rtc::MessageData { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 476 | // We copy the data because the data will become invalid after we |
| 477 | // handle DataMediaChannel::SignalDataReceived but before we fire |
| 478 | // SignalDataReceived. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 479 | DataReceivedMessageData(const ReceiveDataParams& params, |
| 480 | const char* data, |
| 481 | size_t len) |
| 482 | : params(params), payload(data, len) {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 483 | const ReceiveDataParams params; |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 484 | const rtc::CopyOnWriteBuffer payload; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 485 | }; |
| 486 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 487 | typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData; |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 488 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 489 | // overrides from BaseChannel |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 490 | // Checks that data channel type is RTP. |
Harald Alvestrand | 755187f | 2019-12-05 13:43:34 +0100 | [diff] [blame] | 491 | bool CheckDataChannelTypeFromContent(const MediaContentDescription* content, |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 492 | std::string* error_desc); |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 493 | bool SetLocalContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 494 | webrtc::SdpType type, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 495 | std::string* error_desc) override; |
| 496 | bool SetRemoteContent_w(const MediaContentDescription* content, |
Steve Anton | 3828c06 | 2017-12-06 10:34:51 -0800 | [diff] [blame] | 497 | webrtc::SdpType type, |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 498 | std::string* error_desc) override; |
Taylor Brandstetter | bad33bf | 2016-08-25 13:31:14 -0700 | [diff] [blame] | 499 | void UpdateMediaSendRecvState_w() override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 500 | |
Danil Chapovalov | 33b01f2 | 2016-05-11 19:55:27 +0200 | [diff] [blame] | 501 | void OnMessage(rtc::Message* pmsg) override; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 502 | void OnDataReceived(const ReceiveDataParams& params, |
| 503 | const char* data, |
| 504 | size_t len); |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 505 | void OnDataChannelReadyToSend(bool writable); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 506 | |
deadbeef | 953c2ce | 2017-01-09 14:53:41 -0800 | [diff] [blame] | 507 | bool ready_to_send_data_ = false; |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 508 | |
| 509 | // Last DataSendParameters sent down to the media_channel() via |
| 510 | // SetSendParameters. |
| 511 | DataSendParameters last_send_params_; |
| 512 | // Last DataRecvParameters sent down to the media_channel() via |
| 513 | // SetRecvParameters. |
| 514 | DataRecvParameters last_recv_params_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 515 | }; |
| 516 | |
| 517 | } // namespace cricket |
| 518 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 519 | #endif // PC_CHANNEL_H_ |