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| 1 /* | |
| 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_MEDIA_BASE_MEDIACHANNEL_H_ | |
| 29 #define TALK_MEDIA_BASE_MEDIACHANNEL_H_ | |
| 30 | |
| 31 #include <string> | |
| 32 #include <vector> | |
| 33 | |
| 34 #include "talk/media/base/codec.h" | |
| 35 #include "talk/media/base/constants.h" | |
| 36 #include "talk/media/base/streamparams.h" | |
| 37 #include "webrtc/base/basictypes.h" | |
| 38 #include "webrtc/base/buffer.h" | |
| 39 #include "webrtc/base/dscp.h" | |
| 40 #include "webrtc/base/logging.h" | |
| 41 #include "webrtc/base/optional.h" | |
| 42 #include "webrtc/base/sigslot.h" | |
| 43 #include "webrtc/base/socket.h" | |
| 44 #include "webrtc/base/window.h" | |
| 45 #include "webrtc/media/base/videosinkinterface.h" | |
| 46 // TODO(juberti): re-evaluate this include | |
| 47 #include "talk/session/media/audiomonitor.h" | |
| 48 | |
| 49 namespace rtc { | |
| 50 class Buffer; | |
| 51 class RateLimiter; | |
| 52 class Timing; | |
| 53 } | |
| 54 | |
| 55 namespace webrtc { | |
| 56 class AudioSinkInterface; | |
| 57 } | |
| 58 | |
| 59 namespace cricket { | |
| 60 | |
| 61 class AudioRenderer; | |
| 62 class ScreencastId; | |
| 63 class VideoCapturer; | |
| 64 class VideoFrame; | |
| 65 struct RtpHeader; | |
| 66 struct VideoFormat; | |
| 67 | |
| 68 const int kMinRtpHeaderExtensionId = 1; | |
| 69 const int kMaxRtpHeaderExtensionId = 255; | |
| 70 const int kScreencastDefaultFps = 5; | |
| 71 | |
| 72 template <class T> | |
| 73 static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) { | |
| 74 std::string str; | |
| 75 if (val) { | |
| 76 str = key; | |
| 77 str += ": "; | |
| 78 str += val ? rtc::ToString(*val) : ""; | |
| 79 str += ", "; | |
| 80 } | |
| 81 return str; | |
| 82 } | |
| 83 | |
| 84 template <class T> | |
| 85 static std::string VectorToString(const std::vector<T>& vals) { | |
| 86 std::ostringstream ost; | |
| 87 ost << "["; | |
| 88 for (size_t i = 0; i < vals.size(); ++i) { | |
| 89 if (i > 0) { | |
| 90 ost << ", "; | |
| 91 } | |
| 92 ost << vals[i].ToString(); | |
| 93 } | |
| 94 ost << "]"; | |
| 95 return ost.str(); | |
| 96 } | |
| 97 | |
| 98 // Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine. | |
| 99 // Used to be flags, but that makes it hard to selectively apply options. | |
| 100 // We are moving all of the setting of options to structs like this, | |
| 101 // but some things currently still use flags. | |
| 102 struct AudioOptions { | |
| 103 void SetAll(const AudioOptions& change) { | |
| 104 SetFrom(&echo_cancellation, change.echo_cancellation); | |
| 105 SetFrom(&auto_gain_control, change.auto_gain_control); | |
| 106 SetFrom(&noise_suppression, change.noise_suppression); | |
| 107 SetFrom(&highpass_filter, change.highpass_filter); | |
| 108 SetFrom(&stereo_swapping, change.stereo_swapping); | |
| 109 SetFrom(&audio_jitter_buffer_max_packets, | |
| 110 change.audio_jitter_buffer_max_packets); | |
| 111 SetFrom(&audio_jitter_buffer_fast_accelerate, | |
| 112 change.audio_jitter_buffer_fast_accelerate); | |
| 113 SetFrom(&typing_detection, change.typing_detection); | |
| 114 SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise); | |
| 115 SetFrom(&conference_mode, change.conference_mode); | |
| 116 SetFrom(&adjust_agc_delta, change.adjust_agc_delta); | |
| 117 SetFrom(&experimental_agc, change.experimental_agc); | |
| 118 SetFrom(&extended_filter_aec, change.extended_filter_aec); | |
| 119 SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec); | |
| 120 SetFrom(&experimental_ns, change.experimental_ns); | |
| 121 SetFrom(&aec_dump, change.aec_dump); | |
| 122 SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov); | |
| 123 SetFrom(&tx_agc_digital_compression_gain, | |
| 124 change.tx_agc_digital_compression_gain); | |
| 125 SetFrom(&tx_agc_limiter, change.tx_agc_limiter); | |
| 126 SetFrom(&recording_sample_rate, change.recording_sample_rate); | |
| 127 SetFrom(&playout_sample_rate, change.playout_sample_rate); | |
| 128 SetFrom(&dscp, change.dscp); | |
| 129 SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe); | |
| 130 } | |
| 131 | |
| 132 bool operator==(const AudioOptions& o) const { | |
| 133 return echo_cancellation == o.echo_cancellation && | |
| 134 auto_gain_control == o.auto_gain_control && | |
| 135 noise_suppression == o.noise_suppression && | |
| 136 highpass_filter == o.highpass_filter && | |
| 137 stereo_swapping == o.stereo_swapping && | |
| 138 audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets && | |
| 139 audio_jitter_buffer_fast_accelerate == | |
| 140 o.audio_jitter_buffer_fast_accelerate && | |
| 141 typing_detection == o.typing_detection && | |
| 142 aecm_generate_comfort_noise == o.aecm_generate_comfort_noise && | |
| 143 conference_mode == o.conference_mode && | |
| 144 experimental_agc == o.experimental_agc && | |
| 145 extended_filter_aec == o.extended_filter_aec && | |
| 146 delay_agnostic_aec == o.delay_agnostic_aec && | |
| 147 experimental_ns == o.experimental_ns && | |
| 148 adjust_agc_delta == o.adjust_agc_delta && | |
| 149 aec_dump == o.aec_dump && | |
| 150 tx_agc_target_dbov == o.tx_agc_target_dbov && | |
| 151 tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain && | |
| 152 tx_agc_limiter == o.tx_agc_limiter && | |
| 153 recording_sample_rate == o.recording_sample_rate && | |
| 154 playout_sample_rate == o.playout_sample_rate && | |
| 155 dscp == o.dscp && | |
| 156 combined_audio_video_bwe == o.combined_audio_video_bwe; | |
| 157 } | |
| 158 | |
| 159 std::string ToString() const { | |
| 160 std::ostringstream ost; | |
| 161 ost << "AudioOptions {"; | |
| 162 ost << ToStringIfSet("aec", echo_cancellation); | |
| 163 ost << ToStringIfSet("agc", auto_gain_control); | |
| 164 ost << ToStringIfSet("ns", noise_suppression); | |
| 165 ost << ToStringIfSet("hf", highpass_filter); | |
| 166 ost << ToStringIfSet("swap", stereo_swapping); | |
| 167 ost << ToStringIfSet("audio_jitter_buffer_max_packets", | |
| 168 audio_jitter_buffer_max_packets); | |
| 169 ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate", | |
| 170 audio_jitter_buffer_fast_accelerate); | |
| 171 ost << ToStringIfSet("typing", typing_detection); | |
| 172 ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise); | |
| 173 ost << ToStringIfSet("conference", conference_mode); | |
| 174 ost << ToStringIfSet("agc_delta", adjust_agc_delta); | |
| 175 ost << ToStringIfSet("experimental_agc", experimental_agc); | |
| 176 ost << ToStringIfSet("extended_filter_aec", extended_filter_aec); | |
| 177 ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec); | |
| 178 ost << ToStringIfSet("experimental_ns", experimental_ns); | |
| 179 ost << ToStringIfSet("aec_dump", aec_dump); | |
| 180 ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov); | |
| 181 ost << ToStringIfSet("tx_agc_digital_compression_gain", | |
| 182 tx_agc_digital_compression_gain); | |
| 183 ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter); | |
| 184 ost << ToStringIfSet("recording_sample_rate", recording_sample_rate); | |
| 185 ost << ToStringIfSet("playout_sample_rate", playout_sample_rate); | |
| 186 ost << ToStringIfSet("dscp", dscp); | |
| 187 ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe); | |
| 188 ost << "}"; | |
| 189 return ost.str(); | |
| 190 } | |
| 191 | |
| 192 // Audio processing that attempts to filter away the output signal from | |
| 193 // later inbound pickup. | |
| 194 rtc::Optional<bool> echo_cancellation; | |
| 195 // Audio processing to adjust the sensitivity of the local mic dynamically. | |
| 196 rtc::Optional<bool> auto_gain_control; | |
| 197 // Audio processing to filter out background noise. | |
| 198 rtc::Optional<bool> noise_suppression; | |
| 199 // Audio processing to remove background noise of lower frequencies. | |
| 200 rtc::Optional<bool> highpass_filter; | |
| 201 // Audio processing to swap the left and right channels. | |
| 202 rtc::Optional<bool> stereo_swapping; | |
| 203 // Audio receiver jitter buffer (NetEq) max capacity in number of packets. | |
| 204 rtc::Optional<int> audio_jitter_buffer_max_packets; | |
| 205 // Audio receiver jitter buffer (NetEq) fast accelerate mode. | |
| 206 rtc::Optional<bool> audio_jitter_buffer_fast_accelerate; | |
| 207 // Audio processing to detect typing. | |
| 208 rtc::Optional<bool> typing_detection; | |
| 209 rtc::Optional<bool> aecm_generate_comfort_noise; | |
| 210 rtc::Optional<bool> conference_mode; | |
| 211 rtc::Optional<int> adjust_agc_delta; | |
| 212 rtc::Optional<bool> experimental_agc; | |
| 213 rtc::Optional<bool> extended_filter_aec; | |
| 214 rtc::Optional<bool> delay_agnostic_aec; | |
| 215 rtc::Optional<bool> experimental_ns; | |
| 216 rtc::Optional<bool> aec_dump; | |
| 217 // Note that tx_agc_* only applies to non-experimental AGC. | |
| 218 rtc::Optional<uint16_t> tx_agc_target_dbov; | |
| 219 rtc::Optional<uint16_t> tx_agc_digital_compression_gain; | |
| 220 rtc::Optional<bool> tx_agc_limiter; | |
| 221 rtc::Optional<uint32_t> recording_sample_rate; | |
| 222 rtc::Optional<uint32_t> playout_sample_rate; | |
| 223 // Set DSCP value for packet sent from audio channel. | |
| 224 rtc::Optional<bool> dscp; | |
| 225 // Enable combined audio+bandwidth BWE. | |
| 226 rtc::Optional<bool> combined_audio_video_bwe; | |
| 227 | |
| 228 private: | |
| 229 template <typename T> | |
| 230 static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) { | |
| 231 if (o) { | |
| 232 *s = o; | |
| 233 } | |
| 234 } | |
| 235 }; | |
| 236 | |
| 237 // Options that can be applied to a VideoMediaChannel or a VideoMediaEngine. | |
| 238 // Used to be flags, but that makes it hard to selectively apply options. | |
| 239 // We are moving all of the setting of options to structs like this, | |
| 240 // but some things currently still use flags. | |
| 241 struct VideoOptions { | |
| 242 void SetAll(const VideoOptions& change) { | |
| 243 SetFrom(&video_noise_reduction, change.video_noise_reduction); | |
| 244 SetFrom(&cpu_overuse_detection, change.cpu_overuse_detection); | |
| 245 SetFrom(&conference_mode, change.conference_mode); | |
| 246 SetFrom(&dscp, change.dscp); | |
| 247 SetFrom(&suspend_below_min_bitrate, change.suspend_below_min_bitrate); | |
| 248 SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps); | |
| 249 SetFrom(&disable_prerenderer_smoothing, | |
| 250 change.disable_prerenderer_smoothing); | |
| 251 } | |
| 252 | |
| 253 bool operator==(const VideoOptions& o) const { | |
| 254 return video_noise_reduction == o.video_noise_reduction && | |
| 255 cpu_overuse_detection == o.cpu_overuse_detection && | |
| 256 conference_mode == o.conference_mode && | |
| 257 dscp == o.dscp && | |
| 258 suspend_below_min_bitrate == o.suspend_below_min_bitrate && | |
| 259 screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps && | |
| 260 disable_prerenderer_smoothing == o.disable_prerenderer_smoothing; | |
| 261 } | |
| 262 | |
| 263 std::string ToString() const { | |
| 264 std::ostringstream ost; | |
| 265 ost << "VideoOptions {"; | |
| 266 ost << ToStringIfSet("noise reduction", video_noise_reduction); | |
| 267 ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection); | |
| 268 ost << ToStringIfSet("conference mode", conference_mode); | |
| 269 ost << ToStringIfSet("dscp", dscp); | |
| 270 ost << ToStringIfSet("suspend below min bitrate", | |
| 271 suspend_below_min_bitrate); | |
| 272 ost << ToStringIfSet("screencast min bitrate kbps", | |
| 273 screencast_min_bitrate_kbps); | |
| 274 ost << "}"; | |
| 275 return ost.str(); | |
| 276 } | |
| 277 | |
| 278 // Enable denoising? This flag comes from the getUserMedia | |
| 279 // constraint 'googNoiseReduction', and WebRtcVideoEngine2 passes it | |
| 280 // on to the codec options. Disabled by default. | |
| 281 rtc::Optional<bool> video_noise_reduction; | |
| 282 // Enable WebRTC Cpu Overuse Detection. This flag comes from the | |
| 283 // PeerConnection constraint 'googCpuOveruseDetection' and is | |
| 284 // checked in WebRtcVideoChannel2::OnLoadUpdate, where it's passed | |
| 285 // to VideoCapturer::video_adapter()->OnCpuResolutionRequest. | |
| 286 rtc::Optional<bool> cpu_overuse_detection; | |
| 287 // Use conference mode? This flag comes from the remote | |
| 288 // description's SDP line 'a=x-google-flag:conference', copied over | |
| 289 // by VideoChannel::SetRemoteContent_w, and ultimately used by | |
| 290 // conference mode screencast logic in | |
| 291 // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig. | |
| 292 // The special screencast behaviour is disabled by default. | |
| 293 rtc::Optional<bool> conference_mode; | |
| 294 // Set DSCP value for packet sent from video channel. This flag | |
| 295 // comes from the PeerConnection constraint 'googDscp' and, | |
| 296 // WebRtcVideoChannel2::SetOptions checks it before calling | |
| 297 // MediaChannel::SetDscp. If enabled, rtc::DSCP_AF41 is used. If | |
| 298 // disabled, which is the default, rtc::DSCP_DEFAULT is used. | |
| 299 rtc::Optional<bool> dscp; | |
| 300 // Enable WebRTC suspension of video. No video frames will be sent | |
| 301 // when the bitrate is below the configured minimum bitrate. This | |
| 302 // flag comes from the PeerConnection constraint | |
| 303 // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel2 copies it | |
| 304 // to VideoSendStream::Config::suspend_below_min_bitrate. | |
| 305 rtc::Optional<bool> suspend_below_min_bitrate; | |
| 306 // Force screencast to use a minimum bitrate. This flag comes from | |
| 307 // the PeerConnection constraint 'googScreencastMinBitrate'. It is | |
| 308 // copied to the encoder config by WebRtcVideoChannel2. | |
| 309 rtc::Optional<int> screencast_min_bitrate_kbps; | |
| 310 // Set to true if the renderer has an algorithm of frame selection. | |
| 311 // If the value is true, then WebRTC will hand over a frame as soon as | |
| 312 // possible without delay, and rendering smoothness is completely the duty | |
| 313 // of the renderer; | |
| 314 // If the value is false, then WebRTC is responsible to delay frame release | |
| 315 // in order to increase rendering smoothness. | |
| 316 // | |
| 317 // This flag comes from PeerConnection's RtcConfiguration, but is | |
| 318 // currently only set by the command line flag | |
| 319 // 'disable-rtc-smoothness-algorithm'. | |
| 320 // WebRtcVideoChannel2::AddRecvStream copies it to the created | |
| 321 // WebRtcVideoReceiveStream, where it is returned by the | |
| 322 // SmoothsRenderedFrames method. This method is used by the | |
| 323 // VideoReceiveStream, where the value is passed on to the | |
| 324 // IncomingVideoStream constructor. | |
| 325 rtc::Optional<bool> disable_prerenderer_smoothing; | |
| 326 | |
| 327 private: | |
| 328 template <typename T> | |
| 329 static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) { | |
| 330 if (o) { | |
| 331 *s = o; | |
| 332 } | |
| 333 } | |
| 334 }; | |
| 335 | |
| 336 struct RtpHeaderExtension { | |
| 337 RtpHeaderExtension() : id(0) {} | |
| 338 RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {} | |
| 339 | |
| 340 bool operator==(const RtpHeaderExtension& ext) const { | |
| 341 // id is a reserved word in objective-c. Therefore the id attribute has to | |
| 342 // be a fully qualified name in order to compile on IOS. | |
| 343 return this->id == ext.id && | |
| 344 uri == ext.uri; | |
| 345 } | |
| 346 | |
| 347 std::string ToString() const { | |
| 348 std::ostringstream ost; | |
| 349 ost << "{"; | |
| 350 ost << "uri: " << uri; | |
| 351 ost << ", id: " << id; | |
| 352 ost << "}"; | |
| 353 return ost.str(); | |
| 354 } | |
| 355 | |
| 356 std::string uri; | |
| 357 int id; | |
| 358 // TODO(juberti): SendRecv direction; | |
| 359 }; | |
| 360 | |
| 361 // Returns the named header extension if found among all extensions, NULL | |
| 362 // otherwise. | |
| 363 inline const RtpHeaderExtension* FindHeaderExtension( | |
| 364 const std::vector<RtpHeaderExtension>& extensions, | |
| 365 const std::string& name) { | |
| 366 for (std::vector<RtpHeaderExtension>::const_iterator it = extensions.begin(); | |
| 367 it != extensions.end(); ++it) { | |
| 368 if (it->uri == name) | |
| 369 return &(*it); | |
| 370 } | |
| 371 return NULL; | |
| 372 } | |
| 373 | |
| 374 class MediaChannel : public sigslot::has_slots<> { | |
| 375 public: | |
| 376 class NetworkInterface { | |
| 377 public: | |
| 378 enum SocketType { ST_RTP, ST_RTCP }; | |
| 379 virtual bool SendPacket(rtc::Buffer* packet, | |
| 380 const rtc::PacketOptions& options) = 0; | |
| 381 virtual bool SendRtcp(rtc::Buffer* packet, | |
| 382 const rtc::PacketOptions& options) = 0; | |
| 383 virtual int SetOption(SocketType type, rtc::Socket::Option opt, | |
| 384 int option) = 0; | |
| 385 virtual ~NetworkInterface() {} | |
| 386 }; | |
| 387 | |
| 388 MediaChannel() : network_interface_(NULL) {} | |
| 389 virtual ~MediaChannel() {} | |
| 390 | |
| 391 // Sets the abstract interface class for sending RTP/RTCP data. | |
| 392 virtual void SetInterface(NetworkInterface *iface) { | |
| 393 rtc::CritScope cs(&network_interface_crit_); | |
| 394 network_interface_ = iface; | |
| 395 } | |
| 396 | |
| 397 // Called when a RTP packet is received. | |
| 398 virtual void OnPacketReceived(rtc::Buffer* packet, | |
| 399 const rtc::PacketTime& packet_time) = 0; | |
| 400 // Called when a RTCP packet is received. | |
| 401 virtual void OnRtcpReceived(rtc::Buffer* packet, | |
| 402 const rtc::PacketTime& packet_time) = 0; | |
| 403 // Called when the socket's ability to send has changed. | |
| 404 virtual void OnReadyToSend(bool ready) = 0; | |
| 405 // Creates a new outgoing media stream with SSRCs and CNAME as described | |
| 406 // by sp. | |
| 407 virtual bool AddSendStream(const StreamParams& sp) = 0; | |
| 408 // Removes an outgoing media stream. | |
| 409 // ssrc must be the first SSRC of the media stream if the stream uses | |
| 410 // multiple SSRCs. | |
| 411 virtual bool RemoveSendStream(uint32_t ssrc) = 0; | |
| 412 // Creates a new incoming media stream with SSRCs and CNAME as described | |
| 413 // by sp. | |
| 414 virtual bool AddRecvStream(const StreamParams& sp) = 0; | |
| 415 // Removes an incoming media stream. | |
| 416 // ssrc must be the first SSRC of the media stream if the stream uses | |
| 417 // multiple SSRCs. | |
| 418 virtual bool RemoveRecvStream(uint32_t ssrc) = 0; | |
| 419 | |
| 420 // Returns the absoulte sendtime extension id value from media channel. | |
| 421 virtual int GetRtpSendTimeExtnId() const { | |
| 422 return -1; | |
| 423 } | |
| 424 | |
| 425 // Base method to send packet using NetworkInterface. | |
| 426 bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) { | |
| 427 return DoSendPacket(packet, false, options); | |
| 428 } | |
| 429 | |
| 430 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) { | |
| 431 return DoSendPacket(packet, true, options); | |
| 432 } | |
| 433 | |
| 434 int SetOption(NetworkInterface::SocketType type, | |
| 435 rtc::Socket::Option opt, | |
| 436 int option) { | |
| 437 rtc::CritScope cs(&network_interface_crit_); | |
| 438 if (!network_interface_) | |
| 439 return -1; | |
| 440 | |
| 441 return network_interface_->SetOption(type, opt, option); | |
| 442 } | |
| 443 | |
| 444 protected: | |
| 445 // This method sets DSCP |value| on both RTP and RTCP channels. | |
| 446 int SetDscp(rtc::DiffServCodePoint value) { | |
| 447 int ret; | |
| 448 ret = SetOption(NetworkInterface::ST_RTP, | |
| 449 rtc::Socket::OPT_DSCP, | |
| 450 value); | |
| 451 if (ret == 0) { | |
| 452 ret = SetOption(NetworkInterface::ST_RTCP, | |
| 453 rtc::Socket::OPT_DSCP, | |
| 454 value); | |
| 455 } | |
| 456 return ret; | |
| 457 } | |
| 458 | |
| 459 private: | |
| 460 bool DoSendPacket(rtc::Buffer* packet, | |
| 461 bool rtcp, | |
| 462 const rtc::PacketOptions& options) { | |
| 463 rtc::CritScope cs(&network_interface_crit_); | |
| 464 if (!network_interface_) | |
| 465 return false; | |
| 466 | |
| 467 return (!rtcp) ? network_interface_->SendPacket(packet, options) | |
| 468 : network_interface_->SendRtcp(packet, options); | |
| 469 } | |
| 470 | |
| 471 // |network_interface_| can be accessed from the worker_thread and | |
| 472 // from any MediaEngine threads. This critical section is to protect accessing | |
| 473 // of network_interface_ object. | |
| 474 rtc::CriticalSection network_interface_crit_; | |
| 475 NetworkInterface* network_interface_; | |
| 476 }; | |
| 477 | |
| 478 enum SendFlags { | |
| 479 SEND_NOTHING, | |
| 480 SEND_MICROPHONE | |
| 481 }; | |
| 482 | |
| 483 // The stats information is structured as follows: | |
| 484 // Media are represented by either MediaSenderInfo or MediaReceiverInfo. | |
| 485 // Media contains a vector of SSRC infos that are exclusively used by this | |
| 486 // media. (SSRCs shared between media streams can't be represented.) | |
| 487 | |
| 488 // Information about an SSRC. | |
| 489 // This data may be locally recorded, or received in an RTCP SR or RR. | |
| 490 struct SsrcSenderInfo { | |
| 491 SsrcSenderInfo() | |
| 492 : ssrc(0), | |
| 493 timestamp(0) { | |
| 494 } | |
| 495 uint32_t ssrc; | |
| 496 double timestamp; // NTP timestamp, represented as seconds since epoch. | |
| 497 }; | |
| 498 | |
| 499 struct SsrcReceiverInfo { | |
| 500 SsrcReceiverInfo() | |
| 501 : ssrc(0), | |
| 502 timestamp(0) { | |
| 503 } | |
| 504 uint32_t ssrc; | |
| 505 double timestamp; | |
| 506 }; | |
| 507 | |
| 508 struct MediaSenderInfo { | |
| 509 MediaSenderInfo() | |
| 510 : bytes_sent(0), | |
| 511 packets_sent(0), | |
| 512 packets_lost(0), | |
| 513 fraction_lost(0.0), | |
| 514 rtt_ms(0) { | |
| 515 } | |
| 516 void add_ssrc(const SsrcSenderInfo& stat) { | |
| 517 local_stats.push_back(stat); | |
| 518 } | |
| 519 // Temporary utility function for call sites that only provide SSRC. | |
| 520 // As more info is added into SsrcSenderInfo, this function should go away. | |
| 521 void add_ssrc(uint32_t ssrc) { | |
| 522 SsrcSenderInfo stat; | |
| 523 stat.ssrc = ssrc; | |
| 524 add_ssrc(stat); | |
| 525 } | |
| 526 // Utility accessor for clients that are only interested in ssrc numbers. | |
| 527 std::vector<uint32_t> ssrcs() const { | |
| 528 std::vector<uint32_t> retval; | |
| 529 for (std::vector<SsrcSenderInfo>::const_iterator it = local_stats.begin(); | |
| 530 it != local_stats.end(); ++it) { | |
| 531 retval.push_back(it->ssrc); | |
| 532 } | |
| 533 return retval; | |
| 534 } | |
| 535 // Utility accessor for clients that make the assumption only one ssrc | |
| 536 // exists per media. | |
| 537 // This will eventually go away. | |
| 538 uint32_t ssrc() const { | |
| 539 if (local_stats.size() > 0) { | |
| 540 return local_stats[0].ssrc; | |
| 541 } else { | |
| 542 return 0; | |
| 543 } | |
| 544 } | |
| 545 int64_t bytes_sent; | |
| 546 int packets_sent; | |
| 547 int packets_lost; | |
| 548 float fraction_lost; | |
| 549 int64_t rtt_ms; | |
| 550 std::string codec_name; | |
| 551 std::vector<SsrcSenderInfo> local_stats; | |
| 552 std::vector<SsrcReceiverInfo> remote_stats; | |
| 553 }; | |
| 554 | |
| 555 template<class T> | |
| 556 struct VariableInfo { | |
| 557 VariableInfo() | |
| 558 : min_val(), | |
| 559 mean(0.0), | |
| 560 max_val(), | |
| 561 variance(0.0) { | |
| 562 } | |
| 563 T min_val; | |
| 564 double mean; | |
| 565 T max_val; | |
| 566 double variance; | |
| 567 }; | |
| 568 | |
| 569 struct MediaReceiverInfo { | |
| 570 MediaReceiverInfo() | |
| 571 : bytes_rcvd(0), | |
| 572 packets_rcvd(0), | |
| 573 packets_lost(0), | |
| 574 fraction_lost(0.0) { | |
| 575 } | |
| 576 void add_ssrc(const SsrcReceiverInfo& stat) { | |
| 577 local_stats.push_back(stat); | |
| 578 } | |
| 579 // Temporary utility function for call sites that only provide SSRC. | |
| 580 // As more info is added into SsrcSenderInfo, this function should go away. | |
| 581 void add_ssrc(uint32_t ssrc) { | |
| 582 SsrcReceiverInfo stat; | |
| 583 stat.ssrc = ssrc; | |
| 584 add_ssrc(stat); | |
| 585 } | |
| 586 std::vector<uint32_t> ssrcs() const { | |
| 587 std::vector<uint32_t> retval; | |
| 588 for (std::vector<SsrcReceiverInfo>::const_iterator it = local_stats.begin(); | |
| 589 it != local_stats.end(); ++it) { | |
| 590 retval.push_back(it->ssrc); | |
| 591 } | |
| 592 return retval; | |
| 593 } | |
| 594 // Utility accessor for clients that make the assumption only one ssrc | |
| 595 // exists per media. | |
| 596 // This will eventually go away. | |
| 597 uint32_t ssrc() const { | |
| 598 if (local_stats.size() > 0) { | |
| 599 return local_stats[0].ssrc; | |
| 600 } else { | |
| 601 return 0; | |
| 602 } | |
| 603 } | |
| 604 | |
| 605 int64_t bytes_rcvd; | |
| 606 int packets_rcvd; | |
| 607 int packets_lost; | |
| 608 float fraction_lost; | |
| 609 std::string codec_name; | |
| 610 std::vector<SsrcReceiverInfo> local_stats; | |
| 611 std::vector<SsrcSenderInfo> remote_stats; | |
| 612 }; | |
| 613 | |
| 614 struct VoiceSenderInfo : public MediaSenderInfo { | |
| 615 VoiceSenderInfo() | |
| 616 : ext_seqnum(0), | |
| 617 jitter_ms(0), | |
| 618 audio_level(0), | |
| 619 aec_quality_min(0.0), | |
| 620 echo_delay_median_ms(0), | |
| 621 echo_delay_std_ms(0), | |
| 622 echo_return_loss(0), | |
| 623 echo_return_loss_enhancement(0), | |
| 624 typing_noise_detected(false) { | |
| 625 } | |
| 626 | |
| 627 int ext_seqnum; | |
| 628 int jitter_ms; | |
| 629 int audio_level; | |
| 630 float aec_quality_min; | |
| 631 int echo_delay_median_ms; | |
| 632 int echo_delay_std_ms; | |
| 633 int echo_return_loss; | |
| 634 int echo_return_loss_enhancement; | |
| 635 bool typing_noise_detected; | |
| 636 }; | |
| 637 | |
| 638 struct VoiceReceiverInfo : public MediaReceiverInfo { | |
| 639 VoiceReceiverInfo() | |
| 640 : ext_seqnum(0), | |
| 641 jitter_ms(0), | |
| 642 jitter_buffer_ms(0), | |
| 643 jitter_buffer_preferred_ms(0), | |
| 644 delay_estimate_ms(0), | |
| 645 audio_level(0), | |
| 646 expand_rate(0), | |
| 647 speech_expand_rate(0), | |
| 648 secondary_decoded_rate(0), | |
| 649 accelerate_rate(0), | |
| 650 preemptive_expand_rate(0), | |
| 651 decoding_calls_to_silence_generator(0), | |
| 652 decoding_calls_to_neteq(0), | |
| 653 decoding_normal(0), | |
| 654 decoding_plc(0), | |
| 655 decoding_cng(0), | |
| 656 decoding_plc_cng(0), | |
| 657 capture_start_ntp_time_ms(-1) {} | |
| 658 | |
| 659 int ext_seqnum; | |
| 660 int jitter_ms; | |
| 661 int jitter_buffer_ms; | |
| 662 int jitter_buffer_preferred_ms; | |
| 663 int delay_estimate_ms; | |
| 664 int audio_level; | |
| 665 // fraction of synthesized audio inserted through expansion. | |
| 666 float expand_rate; | |
| 667 // fraction of synthesized speech inserted through expansion. | |
| 668 float speech_expand_rate; | |
| 669 // fraction of data out of secondary decoding, including FEC and RED. | |
| 670 float secondary_decoded_rate; | |
| 671 // Fraction of data removed through time compression. | |
| 672 float accelerate_rate; | |
| 673 // Fraction of data inserted through time stretching. | |
| 674 float preemptive_expand_rate; | |
| 675 int decoding_calls_to_silence_generator; | |
| 676 int decoding_calls_to_neteq; | |
| 677 int decoding_normal; | |
| 678 int decoding_plc; | |
| 679 int decoding_cng; | |
| 680 int decoding_plc_cng; | |
| 681 // Estimated capture start time in NTP time in ms. | |
| 682 int64_t capture_start_ntp_time_ms; | |
| 683 }; | |
| 684 | |
| 685 struct VideoSenderInfo : public MediaSenderInfo { | |
| 686 VideoSenderInfo() | |
| 687 : packets_cached(0), | |
| 688 firs_rcvd(0), | |
| 689 plis_rcvd(0), | |
| 690 nacks_rcvd(0), | |
| 691 input_frame_width(0), | |
| 692 input_frame_height(0), | |
| 693 send_frame_width(0), | |
| 694 send_frame_height(0), | |
| 695 framerate_input(0), | |
| 696 framerate_sent(0), | |
| 697 nominal_bitrate(0), | |
| 698 preferred_bitrate(0), | |
| 699 adapt_reason(0), | |
| 700 adapt_changes(0), | |
| 701 avg_encode_ms(0), | |
| 702 encode_usage_percent(0) { | |
| 703 } | |
| 704 | |
| 705 std::vector<SsrcGroup> ssrc_groups; | |
| 706 std::string encoder_implementation_name; | |
| 707 int packets_cached; | |
| 708 int firs_rcvd; | |
| 709 int plis_rcvd; | |
| 710 int nacks_rcvd; | |
| 711 int input_frame_width; | |
| 712 int input_frame_height; | |
| 713 int send_frame_width; | |
| 714 int send_frame_height; | |
| 715 int framerate_input; | |
| 716 int framerate_sent; | |
| 717 int nominal_bitrate; | |
| 718 int preferred_bitrate; | |
| 719 int adapt_reason; | |
| 720 int adapt_changes; | |
| 721 int avg_encode_ms; | |
| 722 int encode_usage_percent; | |
| 723 VariableInfo<int> adapt_frame_drops; | |
| 724 VariableInfo<int> effects_frame_drops; | |
| 725 VariableInfo<double> capturer_frame_time; | |
| 726 }; | |
| 727 | |
| 728 struct VideoReceiverInfo : public MediaReceiverInfo { | |
| 729 VideoReceiverInfo() | |
| 730 : packets_concealed(0), | |
| 731 firs_sent(0), | |
| 732 plis_sent(0), | |
| 733 nacks_sent(0), | |
| 734 frame_width(0), | |
| 735 frame_height(0), | |
| 736 framerate_rcvd(0), | |
| 737 framerate_decoded(0), | |
| 738 framerate_output(0), | |
| 739 framerate_render_input(0), | |
| 740 framerate_render_output(0), | |
| 741 decode_ms(0), | |
| 742 max_decode_ms(0), | |
| 743 jitter_buffer_ms(0), | |
| 744 min_playout_delay_ms(0), | |
| 745 render_delay_ms(0), | |
| 746 target_delay_ms(0), | |
| 747 current_delay_ms(0), | |
| 748 capture_start_ntp_time_ms(-1) { | |
| 749 } | |
| 750 | |
| 751 std::vector<SsrcGroup> ssrc_groups; | |
| 752 std::string decoder_implementation_name; | |
| 753 int packets_concealed; | |
| 754 int firs_sent; | |
| 755 int plis_sent; | |
| 756 int nacks_sent; | |
| 757 int frame_width; | |
| 758 int frame_height; | |
| 759 int framerate_rcvd; | |
| 760 int framerate_decoded; | |
| 761 int framerate_output; | |
| 762 // Framerate as sent to the renderer. | |
| 763 int framerate_render_input; | |
| 764 // Framerate that the renderer reports. | |
| 765 int framerate_render_output; | |
| 766 | |
| 767 // All stats below are gathered per-VideoReceiver, but some will be correlated | |
| 768 // across MediaStreamTracks. NOTE(hta): when sinking stats into per-SSRC | |
| 769 // structures, reflect this in the new layout. | |
| 770 | |
| 771 // Current frame decode latency. | |
| 772 int decode_ms; | |
| 773 // Maximum observed frame decode latency. | |
| 774 int max_decode_ms; | |
| 775 // Jitter (network-related) latency. | |
| 776 int jitter_buffer_ms; | |
| 777 // Requested minimum playout latency. | |
| 778 int min_playout_delay_ms; | |
| 779 // Requested latency to account for rendering delay. | |
| 780 int render_delay_ms; | |
| 781 // Target overall delay: network+decode+render, accounting for | |
| 782 // min_playout_delay_ms. | |
| 783 int target_delay_ms; | |
| 784 // Current overall delay, possibly ramping towards target_delay_ms. | |
| 785 int current_delay_ms; | |
| 786 | |
| 787 // Estimated capture start time in NTP time in ms. | |
| 788 int64_t capture_start_ntp_time_ms; | |
| 789 }; | |
| 790 | |
| 791 struct DataSenderInfo : public MediaSenderInfo { | |
| 792 DataSenderInfo() | |
| 793 : ssrc(0) { | |
| 794 } | |
| 795 | |
| 796 uint32_t ssrc; | |
| 797 }; | |
| 798 | |
| 799 struct DataReceiverInfo : public MediaReceiverInfo { | |
| 800 DataReceiverInfo() | |
| 801 : ssrc(0) { | |
| 802 } | |
| 803 | |
| 804 uint32_t ssrc; | |
| 805 }; | |
| 806 | |
| 807 struct BandwidthEstimationInfo { | |
| 808 BandwidthEstimationInfo() | |
| 809 : available_send_bandwidth(0), | |
| 810 available_recv_bandwidth(0), | |
| 811 target_enc_bitrate(0), | |
| 812 actual_enc_bitrate(0), | |
| 813 retransmit_bitrate(0), | |
| 814 transmit_bitrate(0), | |
| 815 bucket_delay(0) { | |
| 816 } | |
| 817 | |
| 818 int available_send_bandwidth; | |
| 819 int available_recv_bandwidth; | |
| 820 int target_enc_bitrate; | |
| 821 int actual_enc_bitrate; | |
| 822 int retransmit_bitrate; | |
| 823 int transmit_bitrate; | |
| 824 int64_t bucket_delay; | |
| 825 }; | |
| 826 | |
| 827 struct VoiceMediaInfo { | |
| 828 void Clear() { | |
| 829 senders.clear(); | |
| 830 receivers.clear(); | |
| 831 } | |
| 832 std::vector<VoiceSenderInfo> senders; | |
| 833 std::vector<VoiceReceiverInfo> receivers; | |
| 834 }; | |
| 835 | |
| 836 struct VideoMediaInfo { | |
| 837 void Clear() { | |
| 838 senders.clear(); | |
| 839 receivers.clear(); | |
| 840 bw_estimations.clear(); | |
| 841 } | |
| 842 std::vector<VideoSenderInfo> senders; | |
| 843 std::vector<VideoReceiverInfo> receivers; | |
| 844 std::vector<BandwidthEstimationInfo> bw_estimations; | |
| 845 }; | |
| 846 | |
| 847 struct DataMediaInfo { | |
| 848 void Clear() { | |
| 849 senders.clear(); | |
| 850 receivers.clear(); | |
| 851 } | |
| 852 std::vector<DataSenderInfo> senders; | |
| 853 std::vector<DataReceiverInfo> receivers; | |
| 854 }; | |
| 855 | |
| 856 struct RtcpParameters { | |
| 857 bool reduced_size = false; | |
| 858 }; | |
| 859 | |
| 860 template <class Codec> | |
| 861 struct RtpParameters { | |
| 862 virtual std::string ToString() const { | |
| 863 std::ostringstream ost; | |
| 864 ost << "{"; | |
| 865 ost << "codecs: " << VectorToString(codecs) << ", "; | |
| 866 ost << "extensions: " << VectorToString(extensions); | |
| 867 ost << "}"; | |
| 868 return ost.str(); | |
| 869 } | |
| 870 | |
| 871 std::vector<Codec> codecs; | |
| 872 std::vector<RtpHeaderExtension> extensions; | |
| 873 // TODO(pthatcher): Add streams. | |
| 874 RtcpParameters rtcp; | |
| 875 }; | |
| 876 | |
| 877 template <class Codec, class Options> | |
| 878 struct RtpSendParameters : RtpParameters<Codec> { | |
| 879 std::string ToString() const override { | |
| 880 std::ostringstream ost; | |
| 881 ost << "{"; | |
| 882 ost << "codecs: " << VectorToString(this->codecs) << ", "; | |
| 883 ost << "extensions: " << VectorToString(this->extensions) << ", "; | |
| 884 ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", "; | |
| 885 ost << "options: " << options.ToString(); | |
| 886 ost << "}"; | |
| 887 return ost.str(); | |
| 888 } | |
| 889 | |
| 890 int max_bandwidth_bps = -1; | |
| 891 Options options; | |
| 892 }; | |
| 893 | |
| 894 struct AudioSendParameters : RtpSendParameters<AudioCodec, AudioOptions> { | |
| 895 }; | |
| 896 | |
| 897 struct AudioRecvParameters : RtpParameters<AudioCodec> { | |
| 898 }; | |
| 899 | |
| 900 class VoiceMediaChannel : public MediaChannel { | |
| 901 public: | |
| 902 enum Error { | |
| 903 ERROR_NONE = 0, // No error. | |
| 904 ERROR_OTHER, // Other errors. | |
| 905 ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open mic. | |
| 906 ERROR_REC_DEVICE_MUTED, // Mic was muted by OS. | |
| 907 ERROR_REC_DEVICE_SILENT, // No background noise picked up. | |
| 908 ERROR_REC_DEVICE_SATURATION, // Mic input is clipping. | |
| 909 ERROR_REC_DEVICE_REMOVED, // Mic was removed while active. | |
| 910 ERROR_REC_RUNTIME_ERROR, // Processing is encountering errors. | |
| 911 ERROR_REC_SRTP_ERROR, // Generic SRTP failure. | |
| 912 ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets. | |
| 913 ERROR_REC_TYPING_NOISE_DETECTED, // Typing noise is detected. | |
| 914 ERROR_PLAY_DEVICE_OPEN_FAILED = 200, // Could not open playout. | |
| 915 ERROR_PLAY_DEVICE_MUTED, // Playout muted by OS. | |
| 916 ERROR_PLAY_DEVICE_REMOVED, // Playout removed while active. | |
| 917 ERROR_PLAY_RUNTIME_ERROR, // Errors in voice processing. | |
| 918 ERROR_PLAY_SRTP_ERROR, // Generic SRTP failure. | |
| 919 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets. | |
| 920 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected. | |
| 921 }; | |
| 922 | |
| 923 VoiceMediaChannel() {} | |
| 924 virtual ~VoiceMediaChannel() {} | |
| 925 virtual bool SetSendParameters(const AudioSendParameters& params) = 0; | |
| 926 virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0; | |
| 927 // Starts or stops playout of received audio. | |
| 928 virtual bool SetPlayout(bool playout) = 0; | |
| 929 // Starts or stops sending (and potentially capture) of local audio. | |
| 930 virtual bool SetSend(SendFlags flag) = 0; | |
| 931 // Configure stream for sending. | |
| 932 virtual bool SetAudioSend(uint32_t ssrc, | |
| 933 bool enable, | |
| 934 const AudioOptions* options, | |
| 935 AudioRenderer* renderer) = 0; | |
| 936 // Gets current energy levels for all incoming streams. | |
| 937 virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0; | |
| 938 // Get the current energy level of the stream sent to the speaker. | |
| 939 virtual int GetOutputLevel() = 0; | |
| 940 // Get the time in milliseconds since last recorded keystroke, or negative. | |
| 941 virtual int GetTimeSinceLastTyping() = 0; | |
| 942 // Temporarily exposed field for tuning typing detect options. | |
| 943 virtual void SetTypingDetectionParameters(int time_window, | |
| 944 int cost_per_typing, int reporting_threshold, int penalty_decay, | |
| 945 int type_event_delay) = 0; | |
| 946 // Set speaker output volume of the specified ssrc. | |
| 947 virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0; | |
| 948 // Returns if the telephone-event has been negotiated. | |
| 949 virtual bool CanInsertDtmf() = 0; | |
| 950 // Send a DTMF |event|. The DTMF out-of-band signal will be used. | |
| 951 // The |ssrc| should be either 0 or a valid send stream ssrc. | |
| 952 // The valid value for the |event| are 0 to 15 which corresponding to | |
| 953 // DTMF event 0-9, *, #, A-D. | |
| 954 virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0; | |
| 955 // Gets quality stats for the channel. | |
| 956 virtual bool GetStats(VoiceMediaInfo* info) = 0; | |
| 957 | |
| 958 virtual void SetRawAudioSink( | |
| 959 uint32_t ssrc, | |
| 960 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0; | |
| 961 }; | |
| 962 | |
| 963 struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> { | |
| 964 }; | |
| 965 | |
| 966 struct VideoRecvParameters : RtpParameters<VideoCodec> { | |
| 967 }; | |
| 968 | |
| 969 class VideoMediaChannel : public MediaChannel { | |
| 970 public: | |
| 971 enum Error { | |
| 972 ERROR_NONE = 0, // No error. | |
| 973 ERROR_OTHER, // Other errors. | |
| 974 ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open camera. | |
| 975 ERROR_REC_DEVICE_NO_DEVICE, // No camera. | |
| 976 ERROR_REC_DEVICE_IN_USE, // Device is in already use. | |
| 977 ERROR_REC_DEVICE_REMOVED, // Device is removed. | |
| 978 ERROR_REC_SRTP_ERROR, // Generic sender SRTP failure. | |
| 979 ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets. | |
| 980 ERROR_REC_CPU_MAX_CANT_DOWNGRADE, // Can't downgrade capture anymore. | |
| 981 ERROR_PLAY_SRTP_ERROR = 200, // Generic receiver SRTP failure. | |
| 982 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets. | |
| 983 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected. | |
| 984 }; | |
| 985 | |
| 986 VideoMediaChannel() {} | |
| 987 virtual ~VideoMediaChannel() {} | |
| 988 | |
| 989 virtual bool SetSendParameters(const VideoSendParameters& params) = 0; | |
| 990 virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0; | |
| 991 // Gets the currently set codecs/payload types to be used for outgoing media. | |
| 992 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; | |
| 993 // Starts or stops transmission (and potentially capture) of local video. | |
| 994 virtual bool SetSend(bool send) = 0; | |
| 995 // Configure stream for sending. | |
| 996 virtual bool SetVideoSend(uint32_t ssrc, | |
| 997 bool enable, | |
| 998 const VideoOptions* options) = 0; | |
| 999 // Sets the sink object to be used for the specified stream. | |
| 1000 // If SSRC is 0, the renderer is used for the 'default' stream. | |
| 1001 virtual bool SetSink(uint32_t ssrc, | |
| 1002 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0; | |
| 1003 // If |ssrc| is 0, replace the default capturer (engine capturer) with | |
| 1004 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC. | |
| 1005 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0; | |
| 1006 // Gets quality stats for the channel. | |
| 1007 virtual bool GetStats(VideoMediaInfo* info) = 0; | |
| 1008 }; | |
| 1009 | |
| 1010 enum DataMessageType { | |
| 1011 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID | |
| 1012 // values. | |
| 1013 DMT_NONE = 0, | |
| 1014 DMT_CONTROL = 1, | |
| 1015 DMT_BINARY = 2, | |
| 1016 DMT_TEXT = 3, | |
| 1017 }; | |
| 1018 | |
| 1019 // Info about data received in DataMediaChannel. For use in | |
| 1020 // DataMediaChannel::SignalDataReceived and in all of the signals that | |
| 1021 // signal fires, on up the chain. | |
| 1022 struct ReceiveDataParams { | |
| 1023 // The in-packet stream indentifier. | |
| 1024 // For SCTP, this is really SID, not SSRC. | |
| 1025 uint32_t ssrc; | |
| 1026 // The type of message (binary, text, or control). | |
| 1027 DataMessageType type; | |
| 1028 // A per-stream value incremented per packet in the stream. | |
| 1029 int seq_num; | |
| 1030 // A per-stream value monotonically increasing with time. | |
| 1031 int timestamp; | |
| 1032 | |
| 1033 ReceiveDataParams() : | |
| 1034 ssrc(0), | |
| 1035 type(DMT_TEXT), | |
| 1036 seq_num(0), | |
| 1037 timestamp(0) { | |
| 1038 } | |
| 1039 }; | |
| 1040 | |
| 1041 struct SendDataParams { | |
| 1042 // The in-packet stream indentifier. | |
| 1043 // For SCTP, this is really SID, not SSRC. | |
| 1044 uint32_t ssrc; | |
| 1045 // The type of message (binary, text, or control). | |
| 1046 DataMessageType type; | |
| 1047 | |
| 1048 // For SCTP, whether to send messages flagged as ordered or not. | |
| 1049 // If false, messages can be received out of order. | |
| 1050 bool ordered; | |
| 1051 // For SCTP, whether the messages are sent reliably or not. | |
| 1052 // If false, messages may be lost. | |
| 1053 bool reliable; | |
| 1054 // For SCTP, if reliable == false, provide partial reliability by | |
| 1055 // resending up to this many times. Either count or millis | |
| 1056 // is supported, not both at the same time. | |
| 1057 int max_rtx_count; | |
| 1058 // For SCTP, if reliable == false, provide partial reliability by | |
| 1059 // resending for up to this many milliseconds. Either count or millis | |
| 1060 // is supported, not both at the same time. | |
| 1061 int max_rtx_ms; | |
| 1062 | |
| 1063 SendDataParams() : | |
| 1064 ssrc(0), | |
| 1065 type(DMT_TEXT), | |
| 1066 // TODO(pthatcher): Make these true by default? | |
| 1067 ordered(false), | |
| 1068 reliable(false), | |
| 1069 max_rtx_count(0), | |
| 1070 max_rtx_ms(0) { | |
| 1071 } | |
| 1072 }; | |
| 1073 | |
| 1074 enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK }; | |
| 1075 | |
| 1076 struct DataOptions { | |
| 1077 std::string ToString() const { | |
| 1078 return "{}"; | |
| 1079 } | |
| 1080 }; | |
| 1081 | |
| 1082 struct DataSendParameters : RtpSendParameters<DataCodec, DataOptions> { | |
| 1083 std::string ToString() const { | |
| 1084 std::ostringstream ost; | |
| 1085 // Options and extensions aren't used. | |
| 1086 ost << "{"; | |
| 1087 ost << "codecs: " << VectorToString(codecs) << ", "; | |
| 1088 ost << "max_bandwidth_bps: " << max_bandwidth_bps; | |
| 1089 ost << "}"; | |
| 1090 return ost.str(); | |
| 1091 } | |
| 1092 }; | |
| 1093 | |
| 1094 struct DataRecvParameters : RtpParameters<DataCodec> { | |
| 1095 }; | |
| 1096 | |
| 1097 class DataMediaChannel : public MediaChannel { | |
| 1098 public: | |
| 1099 enum Error { | |
| 1100 ERROR_NONE = 0, // No error. | |
| 1101 ERROR_OTHER, // Other errors. | |
| 1102 ERROR_SEND_SRTP_ERROR = 200, // Generic SRTP failure. | |
| 1103 ERROR_SEND_SRTP_AUTH_FAILED, // Failed to authenticate packets. | |
| 1104 ERROR_RECV_SRTP_ERROR, // Generic SRTP failure. | |
| 1105 ERROR_RECV_SRTP_AUTH_FAILED, // Failed to authenticate packets. | |
| 1106 ERROR_RECV_SRTP_REPLAY, // Packet replay detected. | |
| 1107 }; | |
| 1108 | |
| 1109 virtual ~DataMediaChannel() {} | |
| 1110 | |
| 1111 virtual bool SetSendParameters(const DataSendParameters& params) = 0; | |
| 1112 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; | |
| 1113 | |
| 1114 // TODO(pthatcher): Implement this. | |
| 1115 virtual bool GetStats(DataMediaInfo* info) { return true; } | |
| 1116 | |
| 1117 virtual bool SetSend(bool send) = 0; | |
| 1118 virtual bool SetReceive(bool receive) = 0; | |
| 1119 | |
| 1120 virtual bool SendData( | |
| 1121 const SendDataParams& params, | |
| 1122 const rtc::Buffer& payload, | |
| 1123 SendDataResult* result = NULL) = 0; | |
| 1124 // Signals when data is received (params, data, len) | |
| 1125 sigslot::signal3<const ReceiveDataParams&, | |
| 1126 const char*, | |
| 1127 size_t> SignalDataReceived; | |
| 1128 // Signal when the media channel is ready to send the stream. Arguments are: | |
| 1129 // writable(bool) | |
| 1130 sigslot::signal1<bool> SignalReadyToSend; | |
| 1131 // Signal for notifying that the remote side has closed the DataChannel. | |
| 1132 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | |
| 1133 }; | |
| 1134 | |
| 1135 } // namespace cricket | |
| 1136 | |
| 1137 #endif // TALK_MEDIA_BASE_MEDIACHANNEL_H_ | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1587193006:
Move talk/media to webrtc/media (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master