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28 #ifndef TALK_MEDIA_WEBRTCVOICEENGINE_H_
29 #define TALK_MEDIA_WEBRTCVOICEENGINE_H_
36 #include "talk/media/base/rtputils.h"
37 #include "talk/media/webrtc/webrtccommon.h"
38 #include "talk/media/webrtc/webrtcexport.h"
39 #include "talk/media/webrtc/webrtcvoe.h"
40 #include "talk/session/media/channel.h"
41 #include "webrtc/base/buffer.h"
42 #include "webrtc/base/byteorder.h"
43 #include "webrtc/base/logging.h"
44 #include "webrtc/base/scoped_ptr.h"
45 #include "webrtc/base/stream.h"
46 #include "webrtc/common.h"
48 #if !defined(LIBPEERCONNECTION_LIB) && \
49 !defined(LIBPEERCONNECTION_IMPLEMENTATION)
50 // If you hit this, then you've tried to include this header from outside
51 // the shared library. An instance of this class must only be created from
52 // within the library that actually implements it. Otherwise use the
53 // WebRtcMediaEngine to construct an instance.
54 #error "Bogus include."
63 // WebRtcSoundclipStream is an adapter object that allows a memory stream to be
64 // passed into WebRtc, and support looping.
65 class WebRtcSoundclipStream : public webrtc::InStream {
67 WebRtcSoundclipStream(const char* buf, size_t len)
68 : mem_(buf, len), loop_(true) {
70 void set_loop(bool loop) { loop_ = loop; }
71 virtual int Read(void* buf, int len);
75 rtc::MemoryStream mem_;
79 // WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
80 // For now we just dump the data.
81 class WebRtcMonitorStream : public webrtc::OutStream {
82 virtual bool Write(const void *buf, int len) {
87 class AudioDeviceModule;
89 class VoETraceWrapper;
92 class WebRtcSoundclipMedia;
93 class WebRtcVoiceMediaChannel;
95 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
96 // It uses the WebRtc VoiceEngine library for audio handling.
97 class WebRtcVoiceEngine
98 : public webrtc::VoiceEngineObserver,
99 public webrtc::TraceCallback,
100 public webrtc::VoEMediaProcess {
103 // Dependency injection for testing.
104 WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
105 VoEWrapper* voe_wrapper_sc,
106 VoETraceWrapper* tracing);
107 ~WebRtcVoiceEngine();
108 bool Init(rtc::Thread* worker_thread);
111 int GetCapabilities();
112 VoiceMediaChannel* CreateChannel();
114 SoundclipMedia* CreateSoundclip();
116 AudioOptions GetOptions() const { return options_; }
117 bool SetOptions(const AudioOptions& options);
118 // Overrides, when set, take precedence over the options on a
119 // per-option basis. For example, if AGC is set in options and AEC
120 // is set in overrides, AGC and AEC will be both be set. Overrides
121 // can also turn off options. For example, if AGC is set to "on" in
122 // options and AGC is set to "off" in overrides, the result is that
123 // AGC will be off until different overrides are applied or until
124 // the overrides are cleared. Only one set of overrides is present
125 // at a time (they do not "stack"). And when the overrides are
126 // cleared, the media engine's state reverts back to the options set
127 // via SetOptions. This allows us to have both "persistent options"
128 // (the normal options) and "temporary options" (overrides).
129 bool SetOptionOverrides(const AudioOptions& options);
130 bool ClearOptionOverrides();
131 bool SetDelayOffset(int offset);
132 bool SetDevices(const Device* in_device, const Device* out_device);
133 bool GetOutputVolume(int* level);
134 bool SetOutputVolume(int level);
136 bool SetLocalMonitor(bool enable);
138 const std::vector<AudioCodec>& codecs();
139 bool FindCodec(const AudioCodec& codec);
140 bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
142 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
144 void SetLogging(int min_sev, const char* filter);
146 bool RegisterProcessor(uint32 ssrc,
147 VoiceProcessor* voice_processor,
148 MediaProcessorDirection direction);
149 bool UnregisterProcessor(uint32 ssrc,
150 VoiceProcessor* voice_processor,
151 MediaProcessorDirection direction);
153 // Method from webrtc::VoEMediaProcess
154 virtual void Process(int channel,
155 webrtc::ProcessingTypes type,
161 // For tracking WebRtc channels. Needed because we have to pause them
162 // all when switching devices.
163 // May only be called by WebRtcVoiceMediaChannel.
164 void RegisterChannel(WebRtcVoiceMediaChannel *channel);
165 void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
167 // May only be called by WebRtcSoundclipMedia.
168 void RegisterSoundclip(WebRtcSoundclipMedia *channel);
169 void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
171 // Called by WebRtcVoiceMediaChannel to set a gain offset from
172 // the default AGC target level.
173 bool AdjustAgcLevel(int delta);
175 VoEWrapper* voe() { return voe_wrapper_.get(); }
176 VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
177 int GetLastEngineError();
179 // Set the external ADMs. This can only be called before Init.
180 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
181 webrtc::AudioDeviceModule* adm_sc);
183 // Starts AEC dump using existing file.
184 bool StartAecDump(rtc::PlatformFile file);
186 // Check whether the supplied trace should be ignored.
187 bool ShouldIgnoreTrace(const std::string& trace);
189 // Create a VoiceEngine Channel.
190 int CreateMediaVoiceChannel();
191 int CreateSoundclipVoiceChannel();
194 typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
195 typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
197 signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
200 void ConstructCodecs();
202 bool EnsureSoundclipEngineInit();
203 void SetTraceFilter(int filter);
204 void SetTraceOptions(const std::string& options);
205 // Applies either options or overrides. Every option that is "set"
206 // will be applied. Every option not "set" will be ignored. This
207 // allows us to selectively turn on and off different options easily
209 bool ApplyOptions(const AudioOptions& options);
210 virtual void Print(webrtc::TraceLevel level, const char* trace, int length);
211 virtual void CallbackOnError(int channel, int errCode);
212 // Given the device type, name, and id, find device id. Return true and
213 // set the output parameter rtc_id if successful.
214 bool FindWebRtcAudioDeviceId(
215 bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
216 bool FindChannelAndSsrc(int channel_num,
217 WebRtcVoiceMediaChannel** channel,
219 bool FindChannelNumFromSsrc(uint32 ssrc,
220 MediaProcessorDirection direction,
222 bool ChangeLocalMonitor(bool enable);
223 bool PauseLocalMonitor();
224 bool ResumeLocalMonitor();
226 bool UnregisterProcessorChannel(MediaProcessorDirection channel_direction,
228 VoiceProcessor* voice_processor,
229 MediaProcessorDirection processor_direction);
231 void StartAecDump(const std::string& filename);
233 int CreateVoiceChannel(VoEWrapper* voe);
235 // When a voice processor registers with the engine, it is connected
236 // to either the Rx or Tx signals, based on the direction parameter.
237 // SignalXXMediaFrame will be invoked for every audio packet.
238 FrameSignal SignalRxMediaFrame;
239 FrameSignal SignalTxMediaFrame;
241 static const int kDefaultLogSeverity = rtc::LS_WARNING;
243 // The primary instance of WebRtc VoiceEngine.
244 rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
245 // A secondary instance, for playing out soundclips (on the 'ring' device).
246 rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
247 bool voe_wrapper_sc_initialized_;
248 rtc::scoped_ptr<VoETraceWrapper> tracing_;
249 // The external audio device manager
250 webrtc::AudioDeviceModule* adm_;
251 webrtc::AudioDeviceModule* adm_sc_;
253 std::string log_options_;
254 bool is_dumping_aec_;
255 std::vector<AudioCodec> codecs_;
256 std::vector<RtpHeaderExtension> rtp_header_extensions_;
257 bool desired_local_monitor_enable_;
258 rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
259 SoundclipList soundclips_;
260 ChannelList channels_;
261 // channels_ can be read from WebRtc callback thread. We need a lock on that
262 // callback as well as the RegisterChannel/UnregisterChannel.
263 rtc::CriticalSection channels_cs_;
264 webrtc::AgcConfig default_agc_config_;
266 webrtc::Config voe_config_;
269 // See SetOptions and SetOptionOverrides for a description of the
270 // difference between options and overrides.
271 // options_ are the base options, which combined with the
272 // option_overrides_, create the current options being used.
273 // options_ is stored so that when option_overrides_ is cleared, we
274 // can restore the options_ without the option_overrides.
275 AudioOptions options_;
276 AudioOptions option_overrides_;
278 // When the media processor registers with the engine, the ssrc is cached
279 // here so that a look up need not be made when the callback is invoked.
280 // This is necessary because the lookup results in mux_channels_cs lock being
281 // held and if a remote participant leaves the hangout at the same time
282 // we hit a deadlock.
283 uint32 tx_processor_ssrc_;
284 uint32 rx_processor_ssrc_;
286 rtc::CriticalSection signal_media_critical_;
288 // Cache received experimental_aec and experimental_ns values, and apply them
289 // in case they are missing in the audio options. We need to do this because
290 // SetExtraOptions() will revert to defaults for options which are not
292 Settable<bool> experimental_aec_;
293 Settable<bool> experimental_ns_;
296 // WebRtcMediaChannel is a class that implements the common WebRtc channel
298 template <class T, class E>
299 class WebRtcMediaChannel : public T, public webrtc::Transport {
301 WebRtcMediaChannel(E *engine, int channel)
302 : engine_(engine), voe_channel_(channel) {}
303 E *engine() { return engine_; }
304 int voe_channel() const { return voe_channel_; }
305 bool valid() const { return voe_channel_ != -1; }
308 // implements Transport interface
309 virtual int SendPacket(int channel, const void *data, int len) {
310 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
311 if (!T::SendPacket(&packet)) {
317 virtual int SendRTCPPacket(int channel, const void *data, int len) {
318 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
319 return T::SendRtcp(&packet) ? len : -1;
327 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
328 // WebRtc Voice Engine.
329 class WebRtcVoiceMediaChannel
330 : public WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine> {
332 explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
333 virtual ~WebRtcVoiceMediaChannel();
334 virtual bool SetOptions(const AudioOptions& options);
335 virtual bool GetOptions(AudioOptions* options) const {
339 virtual bool SetRecvCodecs(const std::vector<AudioCodec> &codecs);
340 virtual bool SetSendCodecs(const std::vector<AudioCodec> &codecs);
341 virtual bool SetRecvRtpHeaderExtensions(
342 const std::vector<RtpHeaderExtension>& extensions);
343 virtual bool SetSendRtpHeaderExtensions(
344 const std::vector<RtpHeaderExtension>& extensions);
345 virtual bool SetPlayout(bool playout);
347 bool ResumePlayout();
348 virtual bool SetSend(SendFlags send);
351 virtual bool AddSendStream(const StreamParams& sp);
352 virtual bool RemoveSendStream(uint32 ssrc);
353 virtual bool AddRecvStream(const StreamParams& sp);
354 virtual bool RemoveRecvStream(uint32 ssrc);
355 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
356 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
357 virtual bool GetActiveStreams(AudioInfo::StreamList* actives);
358 virtual int GetOutputLevel();
359 virtual int GetTimeSinceLastTyping();
360 virtual void SetTypingDetectionParameters(int time_window,
361 int cost_per_typing, int reporting_threshold, int penalty_decay,
362 int type_event_delay);
363 virtual bool SetOutputScaling(uint32 ssrc, double left, double right);
364 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right);
366 virtual bool SetRingbackTone(const char *buf, int len);
367 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
368 virtual bool CanInsertDtmf();
369 virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags);
371 virtual void OnPacketReceived(rtc::Buffer* packet,
372 const rtc::PacketTime& packet_time);
373 virtual void OnRtcpReceived(rtc::Buffer* packet,
374 const rtc::PacketTime& packet_time);
375 virtual void OnReadyToSend(bool ready) {}
376 virtual bool MuteStream(uint32 ssrc, bool on);
377 virtual bool SetStartSendBandwidth(int bps);
378 virtual bool SetMaxSendBandwidth(int bps);
379 virtual bool GetStats(VoiceMediaInfo* info);
380 // Gets last reported error from WebRtc voice engine. This should be only
381 // called in response a failure.
382 virtual void GetLastMediaError(uint32* ssrc,
383 VoiceMediaChannel::Error* error);
384 bool FindSsrc(int channel_num, uint32* ssrc);
385 void OnError(uint32 ssrc, int error);
387 bool sending() const { return send_ != SEND_NOTHING; }
388 int GetReceiveChannelNum(uint32 ssrc);
389 int GetSendChannelNum(uint32 ssrc);
391 bool SetupSharedBandwidthEstimation(webrtc::VideoEngine* vie,
394 int GetLastEngineError() { return engine()->GetLastEngineError(); }
395 int GetOutputLevel(int channel);
396 bool GetRedSendCodec(const AudioCodec& red_codec,
397 const std::vector<AudioCodec>& all_codecs,
398 webrtc::CodecInst* send_codec);
399 bool EnableRtcp(int channel);
400 bool ResetRecvCodecs(int channel);
401 bool SetPlayout(int channel, bool playout);
402 static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
403 static Error WebRtcErrorToChannelError(int err_code);
406 class WebRtcVoiceChannelRenderer;
407 // Map of ssrc to WebRtcVoiceChannelRenderer object. A new object of
408 // WebRtcVoiceChannelRenderer will be created for every new stream and
409 // will be destroyed when the stream goes away.
410 typedef std::map<uint32, WebRtcVoiceChannelRenderer*> ChannelMap;
411 typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
414 void SetNack(int channel, bool nack_enabled);
415 void SetNack(const ChannelMap& channels, bool nack_enabled);
416 bool SetSendCodec(const webrtc::CodecInst& send_codec);
417 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
418 bool ChangePlayout(bool playout);
419 bool ChangeSend(SendFlags send);
420 bool ChangeSend(int channel, SendFlags send);
421 void ConfigureSendChannel(int channel);
422 bool ConfigureRecvChannel(int channel);
423 bool DeleteChannel(int channel);
424 bool InConferenceMode() const {
425 return options_.conference_mode.GetWithDefaultIfUnset(false);
427 bool IsDefaultChannel(int channel_id) const {
428 return channel_id == voe_channel();
430 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
431 bool SetSendBandwidthInternal(int bps);
433 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
434 const RtpHeaderExtension* extension);
435 bool SetupSharedBweOnChannel(int voe_channel);
437 bool SetChannelRecvRtpHeaderExtensions(
439 const std::vector<RtpHeaderExtension>& extensions);
440 bool SetChannelSendRtpHeaderExtensions(
442 const std::vector<RtpHeaderExtension>& extensions);
444 rtc::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
445 std::set<int> ringback_channels_; // channels playing ringback
446 std::vector<AudioCodec> recv_codecs_;
447 std::vector<AudioCodec> send_codecs_;
448 rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
449 bool send_bw_setting_;
451 AudioOptions options_;
453 bool desired_playout_;
456 bool typing_noise_detected_;
457 SendFlags desired_send_;
459 // shared_bwe_vie_ and shared_bwe_vie_channel_ together identifies a WebRTC
460 // VideoEngine channel that this voice channel should forward incoming packets
461 // to for Bandwidth Estimation purposes.
462 webrtc::VideoEngine* shared_bwe_vie_;
463 int shared_bwe_vie_channel_;
465 // send_channels_ contains the channels which are being used for sending.
466 // When the default channel (voe_channel) is used for sending, it is
467 // contained in send_channels_, otherwise not.
468 ChannelMap send_channels_;
469 std::vector<RtpHeaderExtension> send_extensions_;
470 uint32 default_receive_ssrc_;
471 // Note the default channel (voe_channel()) can reside in both
472 // receive_channels_ and send_channels_ in non-conference mode and in that
473 // case it will only be there if a non-zero default_receive_ssrc_ is set.
474 ChannelMap receive_channels_; // for multiple sources
475 // receive_channels_ can be read from WebRtc callback thread. Access from
476 // the WebRtc thread must be synchronized with edits on the worker thread.
477 // Reads on the worker thread are ok.
479 std::vector<RtpHeaderExtension> receive_extensions_;
480 // Do not lock this on the VoE media processor thread; potential for deadlock
482 mutable rtc::CriticalSection receive_channels_cs_;
485 } // namespace cricket
487 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_