blob: 8faefe6d7b570c69185410237f04d68db28a5b8b [file] [log] [blame]
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_SESSION_MEDIA_CHANNEL_H_
29#define TALK_SESSION_MEDIA_CHANNEL_H_
30
31#include <string>
32#include <vector>
deadbeefcbecd352015-09-23 11:50:27 -070033#include <map>
34#include <set>
35#include <utility>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037#include "talk/media/base/mediachannel.h"
38#include "talk/media/base/mediaengine.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039#include "talk/media/base/streamparams.h"
40#include "talk/media/base/videocapturer.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000041#include "talk/session/media/audiomonitor.h"
buildbot@webrtc.org5ee0f052014-05-05 20:18:08 +000042#include "talk/session/media/bundlefilter.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043#include "talk/session/media/mediamonitor.h"
44#include "talk/session/media/mediasession.h"
45#include "talk/session/media/rtcpmuxfilter.h"
46#include "talk/session/media/srtpfilter.h"
Tommif888bb52015-12-12 01:37:01 +010047#include "webrtc/audio/audio_sink.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000048#include "webrtc/base/asyncudpsocket.h"
49#include "webrtc/base/criticalsection.h"
50#include "webrtc/base/network.h"
51#include "webrtc/base/sigslot.h"
52#include "webrtc/base/window.h"
Tommif888bb52015-12-12 01:37:01 +010053#include "webrtc/p2p/base/transportcontroller.h"
54#include "webrtc/p2p/client/socketmonitor.h"
55
56namespace webrtc {
57class AudioSinkInterface;
58} // namespace webrtc
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059
60namespace cricket {
61
62struct CryptoParams;
63class MediaContentDescription;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000064struct ViewRequest;
65
66enum SinkType {
67 SINK_PRE_CRYPTO, // Sink packets before encryption or after decryption.
68 SINK_POST_CRYPTO // Sink packets after encryption or before decryption.
69};
70
71// BaseChannel contains logic common to voice and video, including
solenberg1dd98f32015-09-10 01:57:14 -070072// enable, marshaling calls to a worker thread, and
henrike@webrtc.org28e20752013-07-10 00:45:36 +000073// connection and media monitors.
wu@webrtc.org78187522013-10-07 23:32:02 +000074//
75// WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
76// This is required to avoid a data race between the destructor modifying the
77// vtable, and the media channel's thread using BaseChannel as the
78// NetworkInterface.
79
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080class BaseChannel
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000081 : public rtc::MessageHandler, public sigslot::has_slots<>,
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +000082 public MediaChannel::NetworkInterface,
83 public ConnectionStatsGetter {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084 public:
deadbeefcbecd352015-09-23 11:50:27 -070085 BaseChannel(rtc::Thread* thread,
86 MediaChannel* channel,
87 TransportController* transport_controller,
88 const std::string& content_name,
89 bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090 virtual ~BaseChannel();
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +000091 bool Init();
wu@webrtc.org78187522013-10-07 23:32:02 +000092 // Deinit may be called multiple times and is simply ignored if it's alreay
93 // done.
94 void Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +000095
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000096 rtc::Thread* worker_thread() const { return worker_thread_; }
deadbeefcbecd352015-09-23 11:50:27 -070097 const std::string& content_name() const { return content_name_; }
98 const std::string& transport_name() const { return transport_name_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099 TransportChannel* transport_channel() const {
100 return transport_channel_;
101 }
102 TransportChannel* rtcp_transport_channel() const {
103 return rtcp_transport_channel_;
104 }
105 bool enabled() const { return enabled_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106
107 // This function returns true if we are using SRTP.
108 bool secure() const { return srtp_filter_.IsActive(); }
109 // The following function returns true if we are using
110 // DTLS-based keying. If you turned off SRTP later, however
111 // you could have secure() == false and dtls_secure() == true.
112 bool secure_dtls() const { return dtls_keyed_; }
113 // This function returns true if we require secure channel for call setup.
114 bool secure_required() const { return secure_required_; }
115
116 bool writable() const { return writable_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700118 // Activate RTCP mux, regardless of the state so far. Once
119 // activated, it can not be deactivated, and if the remote
120 // description doesn't support RTCP mux, setting the remote
121 // description will fail.
122 void ActivateRtcpMux();
deadbeefcbecd352015-09-23 11:50:27 -0700123 bool SetTransport(const std::string& transport_name);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000124 bool PushdownLocalDescription(const SessionDescription* local_desc,
125 ContentAction action,
126 std::string* error_desc);
127 bool PushdownRemoteDescription(const SessionDescription* remote_desc,
128 ContentAction action,
129 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000130 // Channel control
131 bool SetLocalContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000132 ContentAction action,
133 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000134 bool SetRemoteContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000135 ContentAction action,
136 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000137
138 bool Enable(bool enable);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000139
140 // Multiplexing
141 bool AddRecvStream(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200142 bool RemoveRecvStream(uint32_t ssrc);
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000143 bool AddSendStream(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200144 bool RemoveSendStream(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000145
146 // Monitoring
147 void StartConnectionMonitor(int cms);
148 void StopConnectionMonitor();
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000149 // For ConnectionStatsGetter, used by ConnectionMonitor
deadbeefcbecd352015-09-23 11:50:27 -0700150 bool GetConnectionStats(ConnectionInfos* infos) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000151
buildbot@webrtc.org5ee0f052014-05-05 20:18:08 +0000152 BundleFilter* bundle_filter() { return &bundle_filter_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000153
154 const std::vector<StreamParams>& local_streams() const {
155 return local_streams_;
156 }
157 const std::vector<StreamParams>& remote_streams() const {
158 return remote_streams_;
159 }
160
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000161 sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure;
162 void SignalDtlsSetupFailure_w(bool rtcp);
163 void SignalDtlsSetupFailure_s(bool rtcp);
164
buildbot@webrtc.org6bfd6192014-05-15 16:15:59 +0000165 // Used for latency measurements.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000166 sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
167
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000168 // Made public for easier testing.
deadbeefcbecd352015-09-23 11:50:27 -0700169 void SetReadyToSend(bool rtcp, bool ready);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000170
guoweis@webrtc.org4f852882015-03-12 20:09:44 +0000171 // Only public for unit tests. Otherwise, consider protected.
rlesterec9d1872015-10-27 14:22:16 -0700172 int SetOption(SocketType type, rtc::Socket::Option o, int val)
173 override;
guoweis@webrtc.org4f852882015-03-12 20:09:44 +0000174
solenberg5b14b422015-10-01 04:10:31 -0700175 SrtpFilter* srtp_filter() { return &srtp_filter_; }
176
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177 protected:
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000178 virtual MediaChannel* media_channel() const { return media_channel_; }
deadbeefcbecd352015-09-23 11:50:27 -0700179 // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
180 // true). Gets the transport channels from |transport_controller_|.
181 bool SetTransport_w(const std::string& transport_name);
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000182 void set_transport_channel(TransportChannel* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000183 void set_rtcp_transport_channel(TransportChannel* transport);
184 bool was_ever_writable() const { return was_ever_writable_; }
185 void set_local_content_direction(MediaContentDirection direction) {
186 local_content_direction_ = direction;
187 }
188 void set_remote_content_direction(MediaContentDirection direction) {
189 remote_content_direction_ = direction;
190 }
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700191 void set_secure_required(bool secure_required) {
192 secure_required_ = secure_required;
193 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000194 bool IsReadyToReceive() const;
195 bool IsReadyToSend() const;
deadbeefcbecd352015-09-23 11:50:27 -0700196 rtc::Thread* signaling_thread() {
197 return transport_controller_->signaling_thread();
198 }
deadbeefcbecd352015-09-23 11:50:27 -0700199 bool rtcp_transport_enabled() const { return rtcp_transport_enabled_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000200
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000201 void ConnectToTransportChannel(TransportChannel* tc);
202 void DisconnectFromTransportChannel(TransportChannel* tc);
203
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 void FlushRtcpMessages();
205
206 // NetworkInterface implementation, called by MediaEngine
rlesterec9d1872015-10-27 14:22:16 -0700207 bool SendPacket(rtc::Buffer* packet,
208 const rtc::PacketOptions& options) override;
209 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options)
210 override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000211
212 // From TransportChannel
213 void OnWritableState(TransportChannel* channel);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000214 virtual void OnChannelRead(TransportChannel* channel,
215 const char* data,
216 size_t len,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000217 const rtc::PacketTime& packet_time,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000218 int flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000219 void OnReadyToSend(TransportChannel* channel);
220
Guo-wei Shieh1218d7a2015-12-05 09:59:56 -0800221 void OnDtlsState(TransportChannel* channel, DtlsTransportState state);
222
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 bool PacketIsRtcp(const TransportChannel* channel, const char* data,
224 size_t len);
stefanc1aeaf02015-10-15 07:26:07 -0700225 bool SendPacket(bool rtcp,
226 rtc::Buffer* packet,
227 const rtc::PacketOptions& options);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000228 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
229 void HandlePacket(bool rtcp, rtc::Buffer* packet,
230 const rtc::PacketTime& packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000231
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000232 void EnableMedia_w();
233 void DisableMedia_w();
deadbeefcbecd352015-09-23 11:50:27 -0700234 void UpdateWritableState_w();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000235 void ChannelWritable_w();
236 void ChannelNotWritable_w();
237 bool AddRecvStream_w(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200238 bool RemoveRecvStream_w(uint32_t ssrc);
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000239 bool AddSendStream_w(const StreamParams& sp);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200240 bool RemoveSendStream_w(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000241 virtual bool ShouldSetupDtlsSrtp() const;
242 // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
243 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
244 bool SetupDtlsSrtp(bool rtcp_channel);
Guo-wei Shieh1218d7a2015-12-05 09:59:56 -0800245 void MaybeSetupDtlsSrtp_w();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000246 // Set the DTLS-SRTP cipher policy on this channel as appropriate.
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800247 bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000248
249 virtual void ChangeState() = 0;
250
251 // Gets the content info appropriate to the channel (audio or video).
252 virtual const ContentInfo* GetFirstContent(
253 const SessionDescription* sdesc) = 0;
254 bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000255 ContentAction action,
256 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000257 bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000258 ContentAction action,
259 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000261 ContentAction action,
262 std::string* error_desc) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000263 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000264 ContentAction action,
265 std::string* error_desc) = 0;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700266 bool SetRtpTransportParameters_w(const MediaContentDescription* content,
267 ContentAction action,
268 ContentSource src,
269 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000270
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000271 // Helper method to get RTP Absoulute SendTime extension header id if
272 // present in remote supported extensions list.
273 void MaybeCacheRtpAbsSendTimeHeaderExtension(
stefanc1aeaf02015-10-15 07:26:07 -0700274 const std::vector<RtpHeaderExtension>& extensions);
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000275
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000276 bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
277 bool* dtls,
278 std::string* error_desc);
279 bool SetSrtp_w(const std::vector<CryptoParams>& params,
280 ContentAction action,
281 ContentSource src,
282 std::string* error_desc);
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700283 void ActivateRtcpMux_w();
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000284 bool SetRtcpMux_w(bool enable,
285 ContentAction action,
286 ContentSource src,
287 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000288
289 // From MessageHandler
rlesterec9d1872015-10-27 14:22:16 -0700290 void OnMessage(rtc::Message* pmsg) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000291
292 // Handled in derived classes
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800293 // Get the SRTP crypto suites to use for RTP media
294 virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000295 virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000296 const std::vector<ConnectionInfo>& infos) = 0;
297
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000298 // Helper function for invoking bool-returning methods on the worker thread.
299 template <class FunctorT>
300 bool InvokeOnWorker(const FunctorT& functor) {
301 return worker_thread_->Invoke<bool>(functor);
302 }
303
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000304 private:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000305 rtc::Thread* worker_thread_;
deadbeefcbecd352015-09-23 11:50:27 -0700306 TransportController* transport_controller_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 MediaChannel* media_channel_;
308 std::vector<StreamParams> local_streams_;
309 std::vector<StreamParams> remote_streams_;
310
pthatcher@webrtc.org990a00c2015-03-13 18:20:33 +0000311 const std::string content_name_;
deadbeefcbecd352015-09-23 11:50:27 -0700312 std::string transport_name_;
313 bool rtcp_transport_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000314 TransportChannel* transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700315 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000316 TransportChannel* rtcp_transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700317 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000318 SrtpFilter srtp_filter_;
319 RtcpMuxFilter rtcp_mux_filter_;
buildbot@webrtc.org5ee0f052014-05-05 20:18:08 +0000320 BundleFilter bundle_filter_;
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000321 rtc::scoped_ptr<ConnectionMonitor> connection_monitor_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000322 bool enabled_;
323 bool writable_;
324 bool rtp_ready_to_send_;
325 bool rtcp_ready_to_send_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000326 bool was_ever_writable_;
327 MediaContentDirection local_content_direction_;
328 MediaContentDirection remote_content_direction_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000329 bool has_received_packet_;
330 bool dtls_keyed_;
331 bool secure_required_;
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000332 int rtp_abs_sendtime_extn_id_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000333};
334
335// VoiceChannel is a specialization that adds support for early media, DTMF,
336// and input/output level monitoring.
337class VoiceChannel : public BaseChannel {
338 public:
deadbeefcbecd352015-09-23 11:50:27 -0700339 VoiceChannel(rtc::Thread* thread,
340 MediaEngineInterface* media_engine,
341 VoiceMediaChannel* channel,
342 TransportController* transport_controller,
343 const std::string& content_name,
344 bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000345 ~VoiceChannel();
346 bool Init();
solenberg1dd98f32015-09-10 01:57:14 -0700347
348 // Configure sending media on the stream with SSRC |ssrc|
349 // If there is only one sending stream SSRC 0 can be used.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200350 bool SetAudioSend(uint32_t ssrc,
solenbergdfc8f4f2015-10-01 02:31:10 -0700351 bool enable,
deadbeefcbecd352015-09-23 11:50:27 -0700352 const AudioOptions* options,
solenberg1dd98f32015-09-10 01:57:14 -0700353 AudioRenderer* renderer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000354
355 // downcasts a MediaChannel
356 virtual VoiceMediaChannel* media_channel() const {
357 return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
358 }
359
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000360 void SetEarlyMedia(bool enable);
361 // This signal is emitted when we have gone a period of time without
362 // receiving early media. When received, a UI should start playing its
363 // own ringing sound
364 sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
365
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000366 // Returns if the telephone-event has been negotiated.
367 bool CanInsertDtmf();
368 // Send and/or play a DTMF |event| according to the |flags|.
369 // The DTMF out-of-band signal will be used on sending.
370 // The |ssrc| should be either 0 or a valid send stream ssrc.
henrike@webrtc.org9de257d2013-07-17 14:42:53 +0000371 // The valid value for the |event| are 0 which corresponding to DTMF
372 // event 0-9, *, #, A-D.
solenberg1d63dd02015-12-02 12:35:09 -0800373 bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
solenberg4bac9c52015-10-09 02:32:53 -0700374 bool SetOutputVolume(uint32_t ssrc, double volume);
Tommif888bb52015-12-12 01:37:01 +0100375 void SetRawAudioSink(uint32_t ssrc,
376 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink);
377
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000378 // Get statistics about the current media session.
379 bool GetStats(VoiceMediaInfo* stats);
380
381 // Monitoring functions
382 sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
383 SignalConnectionMonitor;
384
385 void StartMediaMonitor(int cms);
386 void StopMediaMonitor();
387 sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
388
389 void StartAudioMonitor(int cms);
390 void StopAudioMonitor();
391 bool IsAudioMonitorRunning() const;
392 sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
393
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000394 int GetInputLevel_w();
395 int GetOutputLevel_w();
396 void GetActiveStreams_w(AudioInfo::StreamList* actives);
397
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000398 private:
399 // overrides from BaseChannel
400 virtual void OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000401 const char* data, size_t len,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000402 const rtc::PacketTime& packet_time,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000403 int flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000404 virtual void ChangeState();
405 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
406 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000407 ContentAction action,
408 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000409 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000410 ContentAction action,
411 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000412 void HandleEarlyMediaTimeout();
solenberg1d63dd02015-12-02 12:35:09 -0800413 bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
solenberg4bac9c52015-10-09 02:32:53 -0700414 bool SetOutputVolume_w(uint32_t ssrc, double volume);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000415 bool GetStats_w(VoiceMediaInfo* stats);
416
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000417 virtual void OnMessage(rtc::Message* pmsg);
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800418 virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000419 virtual void OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000420 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000421 virtual void OnMediaMonitorUpdate(
422 VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
423 void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000424
425 static const int kEarlyMediaTimeout = 1000;
Fredrik Solenberg0c022642015-08-05 12:25:22 +0200426 MediaEngineInterface* media_engine_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000427 bool received_media_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000428 rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_;
429 rtc::scoped_ptr<AudioMonitor> audio_monitor_;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700430
431 // Last AudioSendParameters sent down to the media_channel() via
432 // SetSendParameters.
433 AudioSendParameters last_send_params_;
434 // Last AudioRecvParameters sent down to the media_channel() via
435 // SetRecvParameters.
436 AudioRecvParameters last_recv_params_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000437};
438
439// VideoChannel is a specialization for video.
440class VideoChannel : public BaseChannel {
441 public:
deadbeefcbecd352015-09-23 11:50:27 -0700442 VideoChannel(rtc::Thread* thread,
443 VideoMediaChannel* channel,
444 TransportController* transport_controller,
445 const std::string& content_name,
Fredrik Solenberg0c022642015-08-05 12:25:22 +0200446 bool rtcp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000447 ~VideoChannel();
448 bool Init();
449
Fredrik Solenberg4b60c732015-05-07 14:07:48 +0200450 // downcasts a MediaChannel
451 virtual VideoMediaChannel* media_channel() const {
452 return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
453 }
454
Peter Boström0c4e06b2015-10-07 12:23:21 +0200455 bool SetRenderer(uint32_t ssrc, VideoRenderer* renderer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000456 bool ApplyViewRequest(const ViewRequest& request);
457
458 // TODO(pthatcher): Refactor to use a "capture id" instead of an
459 // ssrc here as the "key".
buildbot@webrtc.org65b98d12014-08-07 22:09:08 +0000460 // Passes ownership of the capturer to the channel.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200461 bool AddScreencast(uint32_t ssrc, VideoCapturer* capturer);
462 bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer);
463 bool RemoveScreencast(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000464 // True if we've added a screencast. Doesn't matter if the capturer
465 // has been started or not.
466 bool IsScreencasting();
Peter Boström0c4e06b2015-10-07 12:23:21 +0200467 int GetScreencastFps(uint32_t ssrc);
468 int GetScreencastMaxPixels(uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000469 // Get statistics about the current media session.
pbos@webrtc.org058b1f12015-03-04 08:54:32 +0000470 bool GetStats(VideoMediaInfo* stats);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000471
472 sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
473 SignalConnectionMonitor;
474
475 void StartMediaMonitor(int cms);
476 void StopMediaMonitor();
477 sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200478 sigslot::signal2<uint32_t, rtc::WindowEvent> SignalScreencastWindowEvent;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000479
480 bool SendIntraFrame();
481 bool RequestIntraFrame();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000482
Peter Boström0c4e06b2015-10-07 12:23:21 +0200483 bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000484
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000485 private:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200486 typedef std::map<uint32_t, VideoCapturer*> ScreencastMap;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000487 struct ScreencastDetailsData;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000488
489 // overrides from BaseChannel
490 virtual void ChangeState();
491 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
492 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000493 ContentAction action,
494 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000495 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000496 ContentAction action,
497 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000498 bool ApplyViewRequest_w(const ViewRequest& request);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000499
Peter Boström0c4e06b2015-10-07 12:23:21 +0200500 bool AddScreencast_w(uint32_t ssrc, VideoCapturer* capturer);
501 bool RemoveScreencast_w(uint32_t ssrc);
502 void OnScreencastWindowEvent_s(uint32_t ssrc, rtc::WindowEvent we);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000503 bool IsScreencasting_w() const;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000504 void GetScreencastDetails_w(ScreencastDetailsData* d) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000505 bool GetStats_w(VideoMediaInfo* stats);
506
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000507 virtual void OnMessage(rtc::Message* pmsg);
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800508 virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000509 virtual void OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000510 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000511 virtual void OnMediaMonitorUpdate(
512 VideoMediaChannel* media_channel, const VideoMediaInfo& info);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200513 virtual void OnScreencastWindowEvent(uint32_t ssrc, rtc::WindowEvent event);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000514 virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200515 bool GetLocalSsrc(const VideoCapturer* capturer, uint32_t* ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000516
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000517 VideoRenderer* renderer_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000518 ScreencastMap screencast_capturers_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000519 rtc::scoped_ptr<VideoMediaMonitor> media_monitor_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000520
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000521 rtc::WindowEvent previous_we_;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700522
523 // Last VideoSendParameters sent down to the media_channel() via
524 // SetSendParameters.
525 VideoSendParameters last_send_params_;
526 // Last VideoRecvParameters sent down to the media_channel() via
527 // SetRecvParameters.
528 VideoRecvParameters last_recv_params_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000529};
530
531// DataChannel is a specialization for data.
532class DataChannel : public BaseChannel {
533 public:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000534 DataChannel(rtc::Thread* thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000535 DataMediaChannel* media_channel,
deadbeefcbecd352015-09-23 11:50:27 -0700536 TransportController* transport_controller,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000537 const std::string& content_name,
538 bool rtcp);
539 ~DataChannel();
540 bool Init();
541
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000542 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000543 const rtc::Buffer& payload,
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000544 SendDataResult* result);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000545
546 void StartMediaMonitor(int cms);
547 void StopMediaMonitor();
548
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000549 // Should be called on the signaling thread only.
550 bool ready_to_send_data() const {
551 return ready_to_send_data_;
552 }
553
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000554 sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
555 sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
556 SignalConnectionMonitor;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200557 sigslot::signal3<DataChannel*, const ReceiveDataParams&, const rtc::Buffer&>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000558 SignalDataReceived;
559 // Signal for notifying when the channel becomes ready to send data.
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000560 // That occurs when the channel is enabled, the transport is writable,
561 // both local and remote descriptions are set, and the channel is unblocked.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000562 sigslot::signal1<bool> SignalReadyToSendData;
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +0000563 // Signal for notifying that the remote side has closed the DataChannel.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200564 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000565
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000566 protected:
567 // downcasts a MediaChannel.
568 virtual DataMediaChannel* media_channel() const {
569 return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
570 }
571
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000572 private:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000573 struct SendDataMessageData : public rtc::MessageData {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000574 SendDataMessageData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000575 const rtc::Buffer* payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000576 SendDataResult* result)
577 : params(params),
578 payload(payload),
579 result(result),
580 succeeded(false) {
581 }
582
583 const SendDataParams& params;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000584 const rtc::Buffer* payload;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000585 SendDataResult* result;
586 bool succeeded;
587 };
588
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000589 struct DataReceivedMessageData : public rtc::MessageData {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000590 // We copy the data because the data will become invalid after we
591 // handle DataMediaChannel::SignalDataReceived but before we fire
592 // SignalDataReceived.
593 DataReceivedMessageData(
594 const ReceiveDataParams& params, const char* data, size_t len)
595 : params(params),
596 payload(data, len) {
597 }
598 const ReceiveDataParams params;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000599 const rtc::Buffer payload;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000600 };
601
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000602 typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000603
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000604 // overrides from BaseChannel
605 virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
606 // If data_channel_type_ is DCT_NONE, set it. Otherwise, check that
607 // it's the same as what was set previously. Returns false if it's
608 // set to one type one type and changed to another type later.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000609 bool SetDataChannelType(DataChannelType new_data_channel_type,
610 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000611 // Same as SetDataChannelType, but extracts the type from the
612 // DataContentDescription.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000613 bool SetDataChannelTypeFromContent(const DataContentDescription* content,
614 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000615 virtual bool SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000616 ContentAction action,
617 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000618 virtual bool SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000619 ContentAction action,
620 std::string* error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000621 virtual void ChangeState();
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000622 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000623
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000624 virtual void OnMessage(rtc::Message* pmsg);
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800625 virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000626 virtual void OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000627 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000628 virtual void OnMediaMonitorUpdate(
629 DataMediaChannel* media_channel, const DataMediaInfo& info);
630 virtual bool ShouldSetupDtlsSrtp() const;
631 void OnDataReceived(
632 const ReceiveDataParams& params, const char* data, size_t len);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200633 void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000634 void OnDataChannelReadyToSend(bool writable);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200635 void OnStreamClosedRemotely(uint32_t sid);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000636
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000637 rtc::scoped_ptr<DataMediaMonitor> media_monitor_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000638 // TODO(pthatcher): Make a separate SctpDataChannel and
639 // RtpDataChannel instead of using this.
640 DataChannelType data_channel_type_;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000641 bool ready_to_send_data_;
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700642
643 // Last DataSendParameters sent down to the media_channel() via
644 // SetSendParameters.
645 DataSendParameters last_send_params_;
646 // Last DataRecvParameters sent down to the media_channel() via
647 // SetRecvParameters.
648 DataRecvParameters last_recv_params_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000649};
650
651} // namespace cricket
652
653#endif // TALK_SESSION_MEDIA_CHANNEL_H_