3 * Copyright 2004 Google Inc.
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
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
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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.
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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28 #ifndef TALK_MEDIA_BASE_MEDIACHANNEL_H_
29 #define TALK_MEDIA_BASE_MEDIACHANNEL_H_
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/sigslot.h"
42 #include "webrtc/base/socket.h"
43 #include "webrtc/base/window.h"
44 // TODO(juberti): re-evaluate this include
45 #include "talk/session/media/audiomonitor.h"
62 const int kMinRtpHeaderExtensionId = 1;
63 const int kMaxRtpHeaderExtensionId = 255;
64 const int kScreencastDefaultFps = 5;
65 const int kHighStartBitrate = 1500;
67 // Used in AudioOptions and VideoOptions to signify "unset" values.
71 Settable() : set_(false), val_() {}
72 explicit Settable(T val) : set_(true), val_(val) {}
78 bool Get(T* out) const {
83 T GetWithDefaultIfUnset(const T& default_value) const {
84 return set_ ? val_ : default_value;
87 virtual void Set(T val) {
97 void SetFrom(const Settable<T>& o) {
98 // Set this value based on the value of o, iff o is set. If this value is
99 // set and o is unset, the current value will be unchanged.
106 std::string ToString() const {
107 return set_ ? rtc::ToString(val_) : "";
110 bool operator==(const Settable<T>& o) const {
111 // Equal if both are unset with any value or both set with the same value.
112 return (set_ == o.set_) && (!set_ || (val_ == o.val_));
115 bool operator!=(const Settable<T>& o) const {
116 return !operator==(o);
120 void InitializeValue(const T &val) {
129 class SettablePercent : public Settable<float> {
131 virtual void Set(float val) {
138 Settable<float>::Set(val);
143 static std::string ToStringIfSet(const char* key, const Settable<T>& val) {
148 str += val.ToString();
154 // Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
155 // Used to be flags, but that makes it hard to selectively apply options.
156 // We are moving all of the setting of options to structs like this,
157 // but some things currently still use flags.
158 struct AudioOptions {
159 void SetAll(const AudioOptions& change) {
160 echo_cancellation.SetFrom(change.echo_cancellation);
161 auto_gain_control.SetFrom(change.auto_gain_control);
162 rx_auto_gain_control.SetFrom(change.rx_auto_gain_control);
163 noise_suppression.SetFrom(change.noise_suppression);
164 highpass_filter.SetFrom(change.highpass_filter);
165 stereo_swapping.SetFrom(change.stereo_swapping);
166 typing_detection.SetFrom(change.typing_detection);
167 aecm_generate_comfort_noise.SetFrom(change.aecm_generate_comfort_noise);
168 conference_mode.SetFrom(change.conference_mode);
169 adjust_agc_delta.SetFrom(change.adjust_agc_delta);
170 experimental_agc.SetFrom(change.experimental_agc);
171 experimental_aec.SetFrom(change.experimental_aec);
172 experimental_ns.SetFrom(change.experimental_ns);
173 aec_dump.SetFrom(change.aec_dump);
174 tx_agc_target_dbov.SetFrom(change.tx_agc_target_dbov);
175 tx_agc_digital_compression_gain.SetFrom(
176 change.tx_agc_digital_compression_gain);
177 tx_agc_limiter.SetFrom(change.tx_agc_limiter);
178 rx_agc_target_dbov.SetFrom(change.rx_agc_target_dbov);
179 rx_agc_digital_compression_gain.SetFrom(
180 change.rx_agc_digital_compression_gain);
181 rx_agc_limiter.SetFrom(change.rx_agc_limiter);
182 recording_sample_rate.SetFrom(change.recording_sample_rate);
183 playout_sample_rate.SetFrom(change.playout_sample_rate);
184 dscp.SetFrom(change.dscp);
185 combined_audio_video_bwe.SetFrom(change.combined_audio_video_bwe);
188 bool operator==(const AudioOptions& o) const {
189 return echo_cancellation == o.echo_cancellation &&
190 auto_gain_control == o.auto_gain_control &&
191 rx_auto_gain_control == o.rx_auto_gain_control &&
192 noise_suppression == o.noise_suppression &&
193 highpass_filter == o.highpass_filter &&
194 stereo_swapping == o.stereo_swapping &&
195 typing_detection == o.typing_detection &&
196 aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
197 conference_mode == o.conference_mode &&
198 experimental_agc == o.experimental_agc &&
199 experimental_aec == o.experimental_aec &&
200 experimental_ns == o.experimental_ns &&
201 adjust_agc_delta == o.adjust_agc_delta &&
202 aec_dump == o.aec_dump &&
203 tx_agc_target_dbov == o.tx_agc_target_dbov &&
204 tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
205 tx_agc_limiter == o.tx_agc_limiter &&
206 rx_agc_target_dbov == o.rx_agc_target_dbov &&
207 rx_agc_digital_compression_gain == o.rx_agc_digital_compression_gain &&
208 rx_agc_limiter == o.rx_agc_limiter &&
209 recording_sample_rate == o.recording_sample_rate &&
210 playout_sample_rate == o.playout_sample_rate &&
212 combined_audio_video_bwe == o.combined_audio_video_bwe;
215 std::string ToString() const {
216 std::ostringstream ost;
217 ost << "AudioOptions {";
218 ost << ToStringIfSet("aec", echo_cancellation);
219 ost << ToStringIfSet("agc", auto_gain_control);
220 ost << ToStringIfSet("rx_agc", rx_auto_gain_control);
221 ost << ToStringIfSet("ns", noise_suppression);
222 ost << ToStringIfSet("hf", highpass_filter);
223 ost << ToStringIfSet("swap", stereo_swapping);
224 ost << ToStringIfSet("typing", typing_detection);
225 ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
226 ost << ToStringIfSet("conference", conference_mode);
227 ost << ToStringIfSet("agc_delta", adjust_agc_delta);
228 ost << ToStringIfSet("experimental_agc", experimental_agc);
229 ost << ToStringIfSet("experimental_aec", experimental_aec);
230 ost << ToStringIfSet("experimental_ns", experimental_ns);
231 ost << ToStringIfSet("aec_dump", aec_dump);
232 ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
233 ost << ToStringIfSet("tx_agc_digital_compression_gain",
234 tx_agc_digital_compression_gain);
235 ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
236 ost << ToStringIfSet("rx_agc_target_dbov", rx_agc_target_dbov);
237 ost << ToStringIfSet("rx_agc_digital_compression_gain",
238 rx_agc_digital_compression_gain);
239 ost << ToStringIfSet("rx_agc_limiter", rx_agc_limiter);
240 ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
241 ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
242 ost << ToStringIfSet("dscp", dscp);
243 ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
248 // Audio processing that attempts to filter away the output signal from
249 // later inbound pickup.
250 Settable<bool> echo_cancellation;
251 // Audio processing to adjust the sensitivity of the local mic dynamically.
252 Settable<bool> auto_gain_control;
253 // Audio processing to apply gain to the remote audio.
254 Settable<bool> rx_auto_gain_control;
255 // Audio processing to filter out background noise.
256 Settable<bool> noise_suppression;
257 // Audio processing to remove background noise of lower frequencies.
258 Settable<bool> highpass_filter;
259 // Audio processing to swap the left and right channels.
260 Settable<bool> stereo_swapping;
261 // Audio processing to detect typing.
262 Settable<bool> typing_detection;
263 Settable<bool> aecm_generate_comfort_noise;
264 Settable<bool> conference_mode;
265 Settable<int> adjust_agc_delta;
266 Settable<bool> experimental_agc;
267 Settable<bool> experimental_aec;
268 Settable<bool> experimental_ns;
269 Settable<bool> aec_dump;
270 // Note that tx_agc_* only applies to non-experimental AGC.
271 Settable<uint16> tx_agc_target_dbov;
272 Settable<uint16> tx_agc_digital_compression_gain;
273 Settable<bool> tx_agc_limiter;
274 Settable<uint16> rx_agc_target_dbov;
275 Settable<uint16> rx_agc_digital_compression_gain;
276 Settable<bool> rx_agc_limiter;
277 Settable<uint32> recording_sample_rate;
278 Settable<uint32> playout_sample_rate;
279 // Set DSCP value for packet sent from audio channel.
281 // Enable combined audio+bandwidth BWE.
282 Settable<bool> combined_audio_video_bwe;
285 // Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
286 // Used to be flags, but that makes it hard to selectively apply options.
287 // We are moving all of the setting of options to structs like this,
288 // but some things currently still use flags.
289 struct VideoOptions {
290 enum HighestBitrate {
297 process_adaptation_threshhold.Set(kProcessCpuThreshold);
298 system_low_adaptation_threshhold.Set(kLowSystemCpuThreshold);
299 system_high_adaptation_threshhold.Set(kHighSystemCpuThreshold);
300 unsignalled_recv_stream_limit.Set(kNumDefaultUnsignalledVideoRecvStreams);
303 void SetAll(const VideoOptions& change) {
304 adapt_input_to_cpu_usage.SetFrom(change.adapt_input_to_cpu_usage);
305 adapt_cpu_with_smoothing.SetFrom(change.adapt_cpu_with_smoothing);
306 video_adapt_third.SetFrom(change.video_adapt_third);
307 video_noise_reduction.SetFrom(change.video_noise_reduction);
308 video_start_bitrate.SetFrom(change.video_start_bitrate);
309 video_highest_bitrate.SetFrom(change.video_highest_bitrate);
310 cpu_overuse_detection.SetFrom(change.cpu_overuse_detection);
311 cpu_underuse_threshold.SetFrom(change.cpu_underuse_threshold);
312 cpu_overuse_threshold.SetFrom(change.cpu_overuse_threshold);
313 cpu_underuse_encode_rsd_threshold.SetFrom(
314 change.cpu_underuse_encode_rsd_threshold);
315 cpu_overuse_encode_rsd_threshold.SetFrom(
316 change.cpu_overuse_encode_rsd_threshold);
317 cpu_overuse_encode_usage.SetFrom(change.cpu_overuse_encode_usage);
318 conference_mode.SetFrom(change.conference_mode);
319 process_adaptation_threshhold.SetFrom(change.process_adaptation_threshhold);
320 system_low_adaptation_threshhold.SetFrom(
321 change.system_low_adaptation_threshhold);
322 system_high_adaptation_threshhold.SetFrom(
323 change.system_high_adaptation_threshhold);
324 buffered_mode_latency.SetFrom(change.buffered_mode_latency);
325 dscp.SetFrom(change.dscp);
326 suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
327 unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
328 use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
329 screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
330 use_payload_padding.SetFrom(change.use_payload_padding);
333 bool operator==(const VideoOptions& o) const {
334 return adapt_input_to_cpu_usage == o.adapt_input_to_cpu_usage &&
335 adapt_cpu_with_smoothing == o.adapt_cpu_with_smoothing &&
336 video_adapt_third == o.video_adapt_third &&
337 video_noise_reduction == o.video_noise_reduction &&
338 video_start_bitrate == o.video_start_bitrate &&
339 video_highest_bitrate == o.video_highest_bitrate &&
340 cpu_overuse_detection == o.cpu_overuse_detection &&
341 cpu_underuse_threshold == o.cpu_underuse_threshold &&
342 cpu_overuse_threshold == o.cpu_overuse_threshold &&
343 cpu_underuse_encode_rsd_threshold ==
344 o.cpu_underuse_encode_rsd_threshold &&
345 cpu_overuse_encode_rsd_threshold ==
346 o.cpu_overuse_encode_rsd_threshold &&
347 cpu_overuse_encode_usage == o.cpu_overuse_encode_usage &&
348 conference_mode == o.conference_mode &&
349 process_adaptation_threshhold == o.process_adaptation_threshhold &&
350 system_low_adaptation_threshhold ==
351 o.system_low_adaptation_threshhold &&
352 system_high_adaptation_threshhold ==
353 o.system_high_adaptation_threshhold &&
354 buffered_mode_latency == o.buffered_mode_latency && dscp == o.dscp &&
355 suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
356 unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
357 use_simulcast_adapter == o.use_simulcast_adapter &&
358 screencast_min_bitrate == o.screencast_min_bitrate &&
359 use_payload_padding == o.use_payload_padding;
362 std::string ToString() const {
363 std::ostringstream ost;
364 ost << "VideoOptions {";
365 ost << ToStringIfSet("cpu adaption", adapt_input_to_cpu_usage);
366 ost << ToStringIfSet("cpu adaptation smoothing", adapt_cpu_with_smoothing);
367 ost << ToStringIfSet("video adapt third", video_adapt_third);
368 ost << ToStringIfSet("noise reduction", video_noise_reduction);
369 ost << ToStringIfSet("start bitrate", video_start_bitrate);
370 ost << ToStringIfSet("highest video bitrate", video_highest_bitrate);
371 ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
372 ost << ToStringIfSet("cpu underuse threshold", cpu_underuse_threshold);
373 ost << ToStringIfSet("cpu overuse threshold", cpu_overuse_threshold);
374 ost << ToStringIfSet("cpu underuse encode rsd threshold",
375 cpu_underuse_encode_rsd_threshold);
376 ost << ToStringIfSet("cpu overuse encode rsd threshold",
377 cpu_overuse_encode_rsd_threshold);
378 ost << ToStringIfSet("cpu overuse encode usage",
379 cpu_overuse_encode_usage);
380 ost << ToStringIfSet("conference mode", conference_mode);
381 ost << ToStringIfSet("process", process_adaptation_threshhold);
382 ost << ToStringIfSet("low", system_low_adaptation_threshhold);
383 ost << ToStringIfSet("high", system_high_adaptation_threshhold);
384 ost << ToStringIfSet("buffered mode latency", buffered_mode_latency);
385 ost << ToStringIfSet("dscp", dscp);
386 ost << ToStringIfSet("suspend below min bitrate",
387 suspend_below_min_bitrate);
388 ost << ToStringIfSet("num channels for early receive",
389 unsignalled_recv_stream_limit);
390 ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
391 ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
392 ost << ToStringIfSet("payload padding", use_payload_padding);
397 // Enable CPU adaptation?
398 Settable<bool> adapt_input_to_cpu_usage;
399 // Enable CPU adaptation smoothing?
400 Settable<bool> adapt_cpu_with_smoothing;
401 // Enable video adapt third?
402 Settable<bool> video_adapt_third;
404 Settable<bool> video_noise_reduction;
405 // Experimental: Enable WebRtc higher start bitrate?
406 Settable<int> video_start_bitrate;
407 // Set highest bitrate mode for video.
408 Settable<HighestBitrate> video_highest_bitrate;
409 // Enable WebRTC Cpu Overuse Detection, which is a new version of the CPU
410 // adaptation algorithm. So this option will override the
411 // |adapt_input_to_cpu_usage|.
412 Settable<bool> cpu_overuse_detection;
413 // Low threshold (t1) for cpu overuse adaptation. (Adapt up)
414 // Metric: encode usage (m1). m1 < t1 => underuse.
415 Settable<int> cpu_underuse_threshold;
416 // High threshold (t1) for cpu overuse adaptation. (Adapt down)
417 // Metric: encode usage (m1). m1 > t1 => overuse.
418 Settable<int> cpu_overuse_threshold;
419 // Low threshold (t2) for cpu overuse adaptation. (Adapt up)
420 // Metric: relative standard deviation of encode time (m2).
421 // Optional threshold. If set, (m1 < t1 && m2 < t2) => underuse.
422 // Note: t2 will have no effect if t1 is not set.
423 Settable<int> cpu_underuse_encode_rsd_threshold;
424 // High threshold (t2) for cpu overuse adaptation. (Adapt down)
425 // Metric: relative standard deviation of encode time (m2).
426 // Optional threshold. If set, (m1 > t1 || m2 > t2) => overuse.
427 // Note: t2 will have no effect if t1 is not set.
428 Settable<int> cpu_overuse_encode_rsd_threshold;
429 // Use encode usage for cpu detection.
430 Settable<bool> cpu_overuse_encode_usage;
431 // Use conference mode?
432 Settable<bool> conference_mode;
433 // Threshhold for process cpu adaptation. (Process limit)
434 SettablePercent process_adaptation_threshhold;
435 // Low threshhold for cpu adaptation. (Adapt up)
436 SettablePercent system_low_adaptation_threshhold;
437 // High threshhold for cpu adaptation. (Adapt down)
438 SettablePercent system_high_adaptation_threshhold;
439 // Specify buffered mode latency in milliseconds.
440 Settable<int> buffered_mode_latency;
441 // Set DSCP value for packet sent from video channel.
443 // Enable WebRTC suspension of video. No video frames will be sent when the
444 // bitrate is below the configured minimum bitrate.
445 Settable<bool> suspend_below_min_bitrate;
446 // Limit on the number of early receive channels that can be created.
447 Settable<int> unsignalled_recv_stream_limit;
448 // Enable use of simulcast adapter.
449 Settable<bool> use_simulcast_adapter;
450 // Force screencast to use a minimum bitrate
451 Settable<int> screencast_min_bitrate;
452 // Enable payload padding.
453 Settable<bool> use_payload_padding;
456 // A class for playing out soundclips.
457 class SoundclipMedia {
459 enum SoundclipFlags {
463 virtual ~SoundclipMedia() {}
465 // Plays a sound out to the speakers with the given audio stream. The stream
466 // must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
467 // on this SoundclipMedia, it is stopped. If clip is NULL, nothing is played.
468 // Returns whether it was successful.
469 virtual bool PlaySound(const char *clip, int len, int flags) = 0;
472 struct RtpHeaderExtension {
473 RtpHeaderExtension() : id(0) {}
474 RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}
477 // TODO(juberti): SendRecv direction;
479 bool operator==(const RtpHeaderExtension& ext) const {
480 // id is a reserved word in objective-c. Therefore the id attribute has to
481 // be a fully qualified name in order to compile on IOS.
482 return this->id == ext.id &&
487 // Returns the named header extension if found among all extensions, NULL
489 inline const RtpHeaderExtension* FindHeaderExtension(
490 const std::vector<RtpHeaderExtension>& extensions,
491 const std::string& name) {
492 for (std::vector<RtpHeaderExtension>::const_iterator it = extensions.begin();
493 it != extensions.end(); ++it) {
500 enum MediaChannelOptions {
501 // Tune the stream for conference mode.
502 OPT_CONFERENCE = 0x0001
505 enum VoiceMediaChannelOptions {
506 // Tune the audio stream for vcs with different target levels.
507 OPT_AGC_MINUS_10DB = 0x80000000
510 // DTMF flags to control if a DTMF tone should be played and/or sent.
516 class MediaChannel : public sigslot::has_slots<> {
518 class NetworkInterface {
520 enum SocketType { ST_RTP, ST_RTCP };
521 virtual bool SendPacket(
523 rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
524 virtual bool SendRtcp(
526 rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
527 virtual int SetOption(SocketType type, rtc::Socket::Option opt,
529 virtual ~NetworkInterface() {}
532 MediaChannel() : network_interface_(NULL) {}
533 virtual ~MediaChannel() {}
535 // Sets the abstract interface class for sending RTP/RTCP data.
536 virtual void SetInterface(NetworkInterface *iface) {
537 rtc::CritScope cs(&network_interface_crit_);
538 network_interface_ = iface;
541 // Called when a RTP packet is received.
542 virtual void OnPacketReceived(rtc::Buffer* packet,
543 const rtc::PacketTime& packet_time) = 0;
544 // Called when a RTCP packet is received.
545 virtual void OnRtcpReceived(rtc::Buffer* packet,
546 const rtc::PacketTime& packet_time) = 0;
547 // Called when the socket's ability to send has changed.
548 virtual void OnReadyToSend(bool ready) = 0;
549 // Creates a new outgoing media stream with SSRCs and CNAME as described
551 virtual bool AddSendStream(const StreamParams& sp) = 0;
552 // Removes an outgoing media stream.
553 // ssrc must be the first SSRC of the media stream if the stream uses
555 virtual bool RemoveSendStream(uint32 ssrc) = 0;
556 // Creates a new incoming media stream with SSRCs and CNAME as described
558 virtual bool AddRecvStream(const StreamParams& sp) = 0;
559 // Removes an incoming media stream.
560 // ssrc must be the first SSRC of the media stream if the stream uses
562 virtual bool RemoveRecvStream(uint32 ssrc) = 0;
564 // Mutes the channel.
565 virtual bool MuteStream(uint32 ssrc, bool on) = 0;
567 // Sets the RTP extension headers and IDs to use when sending RTP.
568 virtual bool SetRecvRtpHeaderExtensions(
569 const std::vector<RtpHeaderExtension>& extensions) = 0;
570 virtual bool SetSendRtpHeaderExtensions(
571 const std::vector<RtpHeaderExtension>& extensions) = 0;
572 // Returns the absoulte sendtime extension id value from media channel.
573 virtual int GetRtpSendTimeExtnId() const {
576 // Sets the maximum allowed bandwidth to use when sending data.
577 virtual bool SetMaxSendBandwidth(int bps) = 0;
579 // Base method to send packet using NetworkInterface.
580 bool SendPacket(rtc::Buffer* packet) {
581 return DoSendPacket(packet, false);
584 bool SendRtcp(rtc::Buffer* packet) {
585 return DoSendPacket(packet, true);
588 int SetOption(NetworkInterface::SocketType type,
589 rtc::Socket::Option opt,
591 rtc::CritScope cs(&network_interface_crit_);
592 if (!network_interface_)
595 return network_interface_->SetOption(type, opt, option);
599 // This method sets DSCP |value| on both RTP and RTCP channels.
600 int SetDscp(rtc::DiffServCodePoint value) {
602 ret = SetOption(NetworkInterface::ST_RTP,
603 rtc::Socket::OPT_DSCP,
606 ret = SetOption(NetworkInterface::ST_RTCP,
607 rtc::Socket::OPT_DSCP,
614 bool DoSendPacket(rtc::Buffer* packet, bool rtcp) {
615 rtc::CritScope cs(&network_interface_crit_);
616 if (!network_interface_)
619 return (!rtcp) ? network_interface_->SendPacket(packet) :
620 network_interface_->SendRtcp(packet);
623 // |network_interface_| can be accessed from the worker_thread and
624 // from any MediaEngine threads. This critical section is to protect accessing
625 // of network_interface_ object.
626 rtc::CriticalSection network_interface_crit_;
627 NetworkInterface* network_interface_;
636 // The stats information is structured as follows:
637 // Media are represented by either MediaSenderInfo or MediaReceiverInfo.
638 // Media contains a vector of SSRC infos that are exclusively used by this
639 // media. (SSRCs shared between media streams can't be represented.)
641 // Information about an SSRC.
642 // This data may be locally recorded, or received in an RTCP SR or RR.
643 struct SsrcSenderInfo {
649 double timestamp; // NTP timestamp, represented as seconds since epoch.
652 struct SsrcReceiverInfo {
661 struct MediaSenderInfo {
669 void add_ssrc(const SsrcSenderInfo& stat) {
670 local_stats.push_back(stat);
672 // Temporary utility function for call sites that only provide SSRC.
673 // As more info is added into SsrcSenderInfo, this function should go away.
674 void add_ssrc(uint32 ssrc) {
679 // Utility accessor for clients that are only interested in ssrc numbers.
680 std::vector<uint32> ssrcs() const {
681 std::vector<uint32> retval;
682 for (std::vector<SsrcSenderInfo>::const_iterator it = local_stats.begin();
683 it != local_stats.end(); ++it) {
684 retval.push_back(it->ssrc);
688 // Utility accessor for clients that make the assumption only one ssrc
690 // This will eventually go away.
691 uint32 ssrc() const {
692 if (local_stats.size() > 0) {
693 return local_stats[0].ssrc;
703 std::string codec_name;
704 std::vector<SsrcSenderInfo> local_stats;
705 std::vector<SsrcReceiverInfo> remote_stats;
709 struct VariableInfo {
722 struct MediaReceiverInfo {
729 void add_ssrc(const SsrcReceiverInfo& stat) {
730 local_stats.push_back(stat);
732 // Temporary utility function for call sites that only provide SSRC.
733 // As more info is added into SsrcSenderInfo, this function should go away.
734 void add_ssrc(uint32 ssrc) {
735 SsrcReceiverInfo stat;
739 std::vector<uint32> ssrcs() const {
740 std::vector<uint32> retval;
741 for (std::vector<SsrcReceiverInfo>::const_iterator it = local_stats.begin();
742 it != local_stats.end(); ++it) {
743 retval.push_back(it->ssrc);
747 // Utility accessor for clients that make the assumption only one ssrc
749 // This will eventually go away.
750 uint32 ssrc() const {
751 if (local_stats.size() > 0) {
752 return local_stats[0].ssrc;
762 std::string codec_name;
763 std::vector<SsrcReceiverInfo> local_stats;
764 std::vector<SsrcSenderInfo> remote_stats;
767 struct VoiceSenderInfo : public MediaSenderInfo {
772 aec_quality_min(0.0),
773 echo_delay_median_ms(0),
774 echo_delay_std_ms(0),
776 echo_return_loss_enhancement(0),
777 typing_noise_detected(false) {
783 float aec_quality_min;
784 int echo_delay_median_ms;
785 int echo_delay_std_ms;
786 int echo_return_loss;
787 int echo_return_loss_enhancement;
788 bool typing_noise_detected;
791 struct VoiceReceiverInfo : public MediaReceiverInfo {
796 jitter_buffer_preferred_ms(0),
797 delay_estimate_ms(0),
800 decoding_calls_to_silence_generator(0),
801 decoding_calls_to_neteq(0),
806 capture_start_ntp_time_ms(-1) {
811 int jitter_buffer_ms;
812 int jitter_buffer_preferred_ms;
813 int delay_estimate_ms;
815 // fraction of synthesized speech inserted through pre-emptive expansion
817 int decoding_calls_to_silence_generator;
818 int decoding_calls_to_neteq;
822 int decoding_plc_cng;
823 // Estimated capture start time in NTP time in ms.
824 int64 capture_start_ntp_time_ms;
827 struct VideoSenderInfo : public MediaSenderInfo {
833 input_frame_width(0),
834 input_frame_height(0),
836 send_frame_height(0),
840 preferred_bitrate(0),
843 capture_jitter_ms(0),
845 encode_usage_percent(0),
846 capture_queue_delay_ms_per_s(0) {
849 std::vector<SsrcGroup> ssrc_groups;
854 int input_frame_width;
855 int input_frame_height;
856 int send_frame_width;
857 int send_frame_height;
861 int preferred_bitrate;
864 int capture_jitter_ms;
866 int encode_usage_percent;
867 int capture_queue_delay_ms_per_s;
868 VariableInfo<int> adapt_frame_drops;
869 VariableInfo<int> effects_frame_drops;
870 VariableInfo<double> capturer_frame_time;
873 struct VideoReceiverInfo : public MediaReceiverInfo {
875 : packets_concealed(0),
882 framerate_decoded(0),
884 framerate_render_input(0),
885 framerate_render_output(0),
889 min_playout_delay_ms(0),
893 capture_start_ntp_time_ms(-1) {
896 std::vector<SsrcGroup> ssrc_groups;
897 int packets_concealed;
904 int framerate_decoded;
905 int framerate_output;
906 // Framerate as sent to the renderer.
907 int framerate_render_input;
908 // Framerate that the renderer reports.
909 int framerate_render_output;
911 // All stats below are gathered per-VideoReceiver, but some will be correlated
912 // across MediaStreamTracks. NOTE(hta): when sinking stats into per-SSRC
913 // structures, reflect this in the new layout.
915 // Current frame decode latency.
917 // Maximum observed frame decode latency.
919 // Jitter (network-related) latency.
920 int jitter_buffer_ms;
921 // Requested minimum playout latency.
922 int min_playout_delay_ms;
923 // Requested latency to account for rendering delay.
925 // Target overall delay: network+decode+render, accounting for
926 // min_playout_delay_ms.
928 // Current overall delay, possibly ramping towards target_delay_ms.
929 int current_delay_ms;
931 // Estimated capture start time in NTP time in ms.
932 int64 capture_start_ntp_time_ms;
935 struct DataSenderInfo : public MediaSenderInfo {
943 struct DataReceiverInfo : public MediaReceiverInfo {
951 struct BandwidthEstimationInfo {
952 BandwidthEstimationInfo()
953 : available_send_bandwidth(0),
954 available_recv_bandwidth(0),
955 target_enc_bitrate(0),
956 actual_enc_bitrate(0),
957 retransmit_bitrate(0),
960 total_received_propagation_delta_ms(0) {
963 int available_send_bandwidth;
964 int available_recv_bandwidth;
965 int target_enc_bitrate;
966 int actual_enc_bitrate;
967 int retransmit_bitrate;
968 int transmit_bitrate;
970 // The following stats are only valid when
971 // StatsOptions::include_received_propagation_stats is true.
972 int total_received_propagation_delta_ms;
973 std::vector<int> recent_received_propagation_delta_ms;
974 std::vector<int64_t> recent_received_packet_group_arrival_time_ms;
977 struct VoiceMediaInfo {
982 std::vector<VoiceSenderInfo> senders;
983 std::vector<VoiceReceiverInfo> receivers;
986 struct VideoMediaInfo {
990 bw_estimations.clear();
992 std::vector<VideoSenderInfo> senders;
993 std::vector<VideoReceiverInfo> receivers;
994 std::vector<BandwidthEstimationInfo> bw_estimations;
997 struct DataMediaInfo {
1002 std::vector<DataSenderInfo> senders;
1003 std::vector<DataReceiverInfo> receivers;
1006 struct StatsOptions {
1007 StatsOptions() : include_received_propagation_stats(false) {}
1009 bool include_received_propagation_stats;
1012 class VoiceMediaChannel : public MediaChannel {
1015 ERROR_NONE = 0, // No error.
1016 ERROR_OTHER, // Other errors.
1017 ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open mic.
1018 ERROR_REC_DEVICE_MUTED, // Mic was muted by OS.
1019 ERROR_REC_DEVICE_SILENT, // No background noise picked up.
1020 ERROR_REC_DEVICE_SATURATION, // Mic input is clipping.
1021 ERROR_REC_DEVICE_REMOVED, // Mic was removed while active.
1022 ERROR_REC_RUNTIME_ERROR, // Processing is encountering errors.
1023 ERROR_REC_SRTP_ERROR, // Generic SRTP failure.
1024 ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets.
1025 ERROR_REC_TYPING_NOISE_DETECTED, // Typing noise is detected.
1026 ERROR_PLAY_DEVICE_OPEN_FAILED = 200, // Could not open playout.
1027 ERROR_PLAY_DEVICE_MUTED, // Playout muted by OS.
1028 ERROR_PLAY_DEVICE_REMOVED, // Playout removed while active.
1029 ERROR_PLAY_RUNTIME_ERROR, // Errors in voice processing.
1030 ERROR_PLAY_SRTP_ERROR, // Generic SRTP failure.
1031 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets.
1032 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected.
1035 VoiceMediaChannel() {}
1036 virtual ~VoiceMediaChannel() {}
1037 // Sets the codecs/payload types to be used for incoming media.
1038 virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) = 0;
1039 // Sets the codecs/payload types to be used for outgoing media.
1040 virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs) = 0;
1041 // Starts or stops playout of received audio.
1042 virtual bool SetPlayout(bool playout) = 0;
1043 // Starts or stops sending (and potentially capture) of local audio.
1044 virtual bool SetSend(SendFlags flag) = 0;
1045 // Sets the renderer object to be used for the specified remote audio stream.
1046 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) = 0;
1047 // Sets the renderer object to be used for the specified local audio stream.
1048 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) = 0;
1049 // Gets current energy levels for all incoming streams.
1050 virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
1051 // Get the current energy level of the stream sent to the speaker.
1052 virtual int GetOutputLevel() = 0;
1053 // Get the time in milliseconds since last recorded keystroke, or negative.
1054 virtual int GetTimeSinceLastTyping() = 0;
1055 // Temporarily exposed field for tuning typing detect options.
1056 virtual void SetTypingDetectionParameters(int time_window,
1057 int cost_per_typing, int reporting_threshold, int penalty_decay,
1058 int type_event_delay) = 0;
1059 // Set left and right scale for speaker output volume of the specified ssrc.
1060 virtual bool SetOutputScaling(uint32 ssrc, double left, double right) = 0;
1061 // Get left and right scale for speaker output volume of the specified ssrc.
1062 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) = 0;
1063 // Specifies a ringback tone to be played during call setup.
1064 virtual bool SetRingbackTone(const char *buf, int len) = 0;
1065 // Plays or stops the aforementioned ringback tone
1066 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) = 0;
1067 // Returns if the telephone-event has been negotiated.
1068 virtual bool CanInsertDtmf() { return false; }
1069 // Send and/or play a DTMF |event| according to the |flags|.
1070 // The DTMF out-of-band signal will be used on sending.
1071 // The |ssrc| should be either 0 or a valid send stream ssrc.
1072 // The valid value for the |event| are 0 to 15 which corresponding to
1073 // DTMF event 0-9, *, #, A-D.
1074 virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) = 0;
1075 // Gets quality stats for the channel.
1076 virtual bool GetStats(VoiceMediaInfo* info) = 0;
1077 // Gets last reported error for this media channel.
1078 virtual void GetLastMediaError(uint32* ssrc,
1079 VoiceMediaChannel::Error* error) {
1080 ASSERT(error != NULL);
1081 *error = ERROR_NONE;
1083 // Sets the media options to use.
1084 virtual bool SetOptions(const AudioOptions& options) = 0;
1085 virtual bool GetOptions(AudioOptions* options) const = 0;
1087 // Signal errors from MediaChannel. Arguments are:
1088 // ssrc(uint32), and error(VoiceMediaChannel::Error).
1089 sigslot::signal2<uint32, VoiceMediaChannel::Error> SignalMediaError;
1092 class VideoMediaChannel : public MediaChannel {
1095 ERROR_NONE = 0, // No error.
1096 ERROR_OTHER, // Other errors.
1097 ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open camera.
1098 ERROR_REC_DEVICE_NO_DEVICE, // No camera.
1099 ERROR_REC_DEVICE_IN_USE, // Device is in already use.
1100 ERROR_REC_DEVICE_REMOVED, // Device is removed.
1101 ERROR_REC_SRTP_ERROR, // Generic sender SRTP failure.
1102 ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets.
1103 ERROR_REC_CPU_MAX_CANT_DOWNGRADE, // Can't downgrade capture anymore.
1104 ERROR_PLAY_SRTP_ERROR = 200, // Generic receiver SRTP failure.
1105 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets.
1106 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected.
1109 VideoMediaChannel() : renderer_(NULL) {}
1110 virtual ~VideoMediaChannel() {}
1111 // Sets the codecs/payload types to be used for incoming media.
1112 virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) = 0;
1113 // Sets the codecs/payload types to be used for outgoing media.
1114 virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs) = 0;
1115 // Gets the currently set codecs/payload types to be used for outgoing media.
1116 virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
1117 // Sets the format of a specified outgoing stream.
1118 virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) = 0;
1119 // Starts or stops playout of received video.
1120 virtual bool SetRender(bool render) = 0;
1121 // Starts or stops transmission (and potentially capture) of local video.
1122 virtual bool SetSend(bool send) = 0;
1123 // Sets the renderer object to be used for the specified stream.
1124 // If SSRC is 0, the renderer is used for the 'default' stream.
1125 virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) = 0;
1126 // If |ssrc| is 0, replace the default capturer (engine capturer) with
1127 // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC.
1128 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) = 0;
1129 // Gets quality stats for the channel.
1130 virtual bool GetStats(const StatsOptions& options, VideoMediaInfo* info) = 0;
1131 // This is needed for MediaMonitor to use the same template for voice, video
1132 // and data MediaChannels.
1133 bool GetStats(VideoMediaInfo* info) {
1134 return GetStats(StatsOptions(), info);
1137 // Send an intra frame to the receivers.
1138 virtual bool SendIntraFrame() = 0;
1139 // Reuqest each of the remote senders to send an intra frame.
1140 virtual bool RequestIntraFrame() = 0;
1141 // Sets the media options to use.
1142 virtual bool SetOptions(const VideoOptions& options) = 0;
1143 virtual bool GetOptions(VideoOptions* options) const = 0;
1144 virtual void UpdateAspectRatio(int ratio_w, int ratio_h) = 0;
1146 // Signal errors from MediaChannel. Arguments are:
1147 // ssrc(uint32), and error(VideoMediaChannel::Error).
1148 sigslot::signal2<uint32, Error> SignalMediaError;
1151 VideoRenderer *renderer_;
1154 enum DataMessageType {
1155 // Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID
1163 // Info about data received in DataMediaChannel. For use in
1164 // DataMediaChannel::SignalDataReceived and in all of the signals that
1165 // signal fires, on up the chain.
1166 struct ReceiveDataParams {
1167 // The in-packet stream indentifier.
1168 // For SCTP, this is really SID, not SSRC.
1170 // The type of message (binary, text, or control).
1171 DataMessageType type;
1172 // A per-stream value incremented per packet in the stream.
1174 // A per-stream value monotonically increasing with time.
1177 ReceiveDataParams() :
1185 struct SendDataParams {
1186 // The in-packet stream indentifier.
1187 // For SCTP, this is really SID, not SSRC.
1189 // The type of message (binary, text, or control).
1190 DataMessageType type;
1192 // For SCTP, whether to send messages flagged as ordered or not.
1193 // If false, messages can be received out of order.
1195 // For SCTP, whether the messages are sent reliably or not.
1196 // If false, messages may be lost.
1198 // For SCTP, if reliable == false, provide partial reliability by
1199 // resending up to this many times. Either count or millis
1200 // is supported, not both at the same time.
1202 // For SCTP, if reliable == false, provide partial reliability by
1203 // resending for up to this many milliseconds. Either count or millis
1204 // is supported, not both at the same time.
1210 // TODO(pthatcher): Make these true by default?
1218 enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
1220 class DataMediaChannel : public MediaChannel {
1223 ERROR_NONE = 0, // No error.
1224 ERROR_OTHER, // Other errors.
1225 ERROR_SEND_SRTP_ERROR = 200, // Generic SRTP failure.
1226 ERROR_SEND_SRTP_AUTH_FAILED, // Failed to authenticate packets.
1227 ERROR_RECV_SRTP_ERROR, // Generic SRTP failure.
1228 ERROR_RECV_SRTP_AUTH_FAILED, // Failed to authenticate packets.
1229 ERROR_RECV_SRTP_REPLAY, // Packet replay detected.
1232 virtual ~DataMediaChannel() {}
1234 virtual bool SetSendCodecs(const std::vector<DataCodec>& codecs) = 0;
1235 virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs) = 0;
1237 virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
1238 // TODO(pthatcher): Implement this.
1239 virtual bool GetStats(DataMediaInfo* info) { return true; }
1241 virtual bool SetSend(bool send) = 0;
1242 virtual bool SetReceive(bool receive) = 0;
1244 virtual bool SendData(
1245 const SendDataParams& params,
1246 const rtc::Buffer& payload,
1247 SendDataResult* result = NULL) = 0;
1248 // Signals when data is received (params, data, len)
1249 sigslot::signal3<const ReceiveDataParams&,
1251 size_t> SignalDataReceived;
1252 // Signal errors from MediaChannel. Arguments are:
1253 // ssrc(uint32), and error(DataMediaChannel::Error).
1254 sigslot::signal2<uint32, DataMediaChannel::Error> SignalMediaError;
1255 // Signal when the media channel is ready to send the stream. Arguments are:
1257 sigslot::signal1<bool> SignalReadyToSend;
1258 // Signal for notifying that the remote side has closed the DataChannel.
1259 sigslot::signal1<uint32> SignalStreamClosedRemotely;
1262 } // namespace cricket
1264 #endif // TALK_MEDIA_BASE_MEDIACHANNEL_H_