3 * Copyright 2004 Google Inc.
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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; }
72 virtual int Read(void* buf, int len) OVERRIDE;
73 virtual int Rewind() OVERRIDE;
76 rtc::MemoryStream mem_;
80 // WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
81 // For now we just dump the data.
82 class WebRtcMonitorStream : public webrtc::OutStream {
83 virtual bool Write(const void *buf, int len) OVERRIDE {
88 class AudioDeviceModule;
90 class VoETraceWrapper;
93 class WebRtcSoundclipMedia;
94 class WebRtcVoiceMediaChannel;
96 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
97 // It uses the WebRtc VoiceEngine library for audio handling.
98 class WebRtcVoiceEngine
99 : public webrtc::VoiceEngineObserver,
100 public webrtc::TraceCallback,
101 public webrtc::VoEMediaProcess {
104 // Dependency injection for testing.
105 WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
106 VoEWrapper* voe_wrapper_sc,
107 VoETraceWrapper* tracing);
108 ~WebRtcVoiceEngine();
109 bool Init(rtc::Thread* worker_thread);
112 int GetCapabilities();
113 VoiceMediaChannel* CreateChannel();
115 SoundclipMedia* CreateSoundclip();
117 AudioOptions GetOptions() const { return options_; }
118 bool SetOptions(const AudioOptions& options);
119 // Overrides, when set, take precedence over the options on a
120 // per-option basis. For example, if AGC is set in options and AEC
121 // is set in overrides, AGC and AEC will be both be set. Overrides
122 // can also turn off options. For example, if AGC is set to "on" in
123 // options and AGC is set to "off" in overrides, the result is that
124 // AGC will be off until different overrides are applied or until
125 // the overrides are cleared. Only one set of overrides is present
126 // at a time (they do not "stack"). And when the overrides are
127 // cleared, the media engine's state reverts back to the options set
128 // via SetOptions. This allows us to have both "persistent options"
129 // (the normal options) and "temporary options" (overrides).
130 bool SetOptionOverrides(const AudioOptions& options);
131 bool ClearOptionOverrides();
132 bool SetDelayOffset(int offset);
133 bool SetDevices(const Device* in_device, const Device* out_device);
134 bool GetOutputVolume(int* level);
135 bool SetOutputVolume(int level);
137 bool SetLocalMonitor(bool enable);
139 const std::vector<AudioCodec>& codecs();
140 bool FindCodec(const AudioCodec& codec);
141 bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
143 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
145 void SetLogging(int min_sev, const char* filter);
147 bool RegisterProcessor(uint32 ssrc,
148 VoiceProcessor* voice_processor,
149 MediaProcessorDirection direction);
150 bool UnregisterProcessor(uint32 ssrc,
151 VoiceProcessor* voice_processor,
152 MediaProcessorDirection direction);
154 // Method from webrtc::VoEMediaProcess
155 virtual void Process(int channel,
156 webrtc::ProcessingTypes type,
160 bool is_stereo) OVERRIDE;
162 // For tracking WebRtc channels. Needed because we have to pause them
163 // all when switching devices.
164 // May only be called by WebRtcVoiceMediaChannel.
165 void RegisterChannel(WebRtcVoiceMediaChannel *channel);
166 void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
168 // May only be called by WebRtcSoundclipMedia.
169 void RegisterSoundclip(WebRtcSoundclipMedia *channel);
170 void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
172 // Called by WebRtcVoiceMediaChannel to set a gain offset from
173 // the default AGC target level.
174 bool AdjustAgcLevel(int delta);
176 VoEWrapper* voe() { return voe_wrapper_.get(); }
177 VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
178 int GetLastEngineError();
180 // Set the external ADMs. This can only be called before Init.
181 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
182 webrtc::AudioDeviceModule* adm_sc);
184 // Starts AEC dump using existing file.
185 bool StartAecDump(rtc::PlatformFile file);
187 // Check whether the supplied trace should be ignored.
188 bool ShouldIgnoreTrace(const std::string& trace);
190 // Create a VoiceEngine Channel.
191 int CreateMediaVoiceChannel();
192 int CreateSoundclipVoiceChannel();
195 typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
196 typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
198 signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
201 void ConstructCodecs();
202 bool GetVoeCodec(int index, webrtc::CodecInst& codec);
204 bool EnsureSoundclipEngineInit();
205 void SetTraceFilter(int filter);
206 void SetTraceOptions(const std::string& options);
207 // Applies either options or overrides. Every option that is "set"
208 // will be applied. Every option not "set" will be ignored. This
209 // allows us to selectively turn on and off different options easily
211 bool ApplyOptions(const AudioOptions& options);
213 // webrtc::TraceCallback:
214 virtual void Print(webrtc::TraceLevel level,
216 int length) OVERRIDE;
218 // webrtc::VoiceEngineObserver:
219 virtual void CallbackOnError(int channel, int errCode) OVERRIDE;
221 // Given the device type, name, and id, find device id. Return true and
222 // set the output parameter rtc_id if successful.
223 bool FindWebRtcAudioDeviceId(
224 bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
225 bool FindChannelAndSsrc(int channel_num,
226 WebRtcVoiceMediaChannel** channel,
228 bool FindChannelNumFromSsrc(uint32 ssrc,
229 MediaProcessorDirection direction,
231 bool ChangeLocalMonitor(bool enable);
232 bool PauseLocalMonitor();
233 bool ResumeLocalMonitor();
235 bool UnregisterProcessorChannel(MediaProcessorDirection channel_direction,
237 VoiceProcessor* voice_processor,
238 MediaProcessorDirection processor_direction);
240 void StartAecDump(const std::string& filename);
242 int CreateVoiceChannel(VoEWrapper* voe);
244 // When a voice processor registers with the engine, it is connected
245 // to either the Rx or Tx signals, based on the direction parameter.
246 // SignalXXMediaFrame will be invoked for every audio packet.
247 FrameSignal SignalRxMediaFrame;
248 FrameSignal SignalTxMediaFrame;
250 static const int kDefaultLogSeverity = rtc::LS_WARNING;
252 // The primary instance of WebRtc VoiceEngine.
253 rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
254 // A secondary instance, for playing out soundclips (on the 'ring' device).
255 rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
256 bool voe_wrapper_sc_initialized_;
257 rtc::scoped_ptr<VoETraceWrapper> tracing_;
258 // The external audio device manager
259 webrtc::AudioDeviceModule* adm_;
260 webrtc::AudioDeviceModule* adm_sc_;
262 std::string log_options_;
263 bool is_dumping_aec_;
264 std::vector<AudioCodec> codecs_;
265 std::vector<RtpHeaderExtension> rtp_header_extensions_;
266 bool desired_local_monitor_enable_;
267 rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
268 SoundclipList soundclips_;
269 ChannelList channels_;
270 // channels_ can be read from WebRtc callback thread. We need a lock on that
271 // callback as well as the RegisterChannel/UnregisterChannel.
272 rtc::CriticalSection channels_cs_;
273 webrtc::AgcConfig default_agc_config_;
275 webrtc::Config voe_config_;
278 // See SetOptions and SetOptionOverrides for a description of the
279 // difference between options and overrides.
280 // options_ are the base options, which combined with the
281 // option_overrides_, create the current options being used.
282 // options_ is stored so that when option_overrides_ is cleared, we
283 // can restore the options_ without the option_overrides.
284 AudioOptions options_;
285 AudioOptions option_overrides_;
287 // When the media processor registers with the engine, the ssrc is cached
288 // here so that a look up need not be made when the callback is invoked.
289 // This is necessary because the lookup results in mux_channels_cs lock being
290 // held and if a remote participant leaves the hangout at the same time
291 // we hit a deadlock.
292 uint32 tx_processor_ssrc_;
293 uint32 rx_processor_ssrc_;
295 rtc::CriticalSection signal_media_critical_;
297 // Cache received experimental_aec and experimental_ns values, and apply them
298 // in case they are missing in the audio options. We need to do this because
299 // SetExtraOptions() will revert to defaults for options which are not
301 Settable<bool> experimental_aec_;
302 Settable<bool> experimental_ns_;
305 // WebRtcMediaChannel is a class that implements the common WebRtc channel
307 template <class T, class E>
308 class WebRtcMediaChannel : public T, public webrtc::Transport {
310 WebRtcMediaChannel(E *engine, int channel)
311 : engine_(engine), voe_channel_(channel) {}
312 E *engine() { return engine_; }
313 int voe_channel() const { return voe_channel_; }
314 bool valid() const { return voe_channel_ != -1; }
317 // implements Transport interface
318 virtual int SendPacket(int channel, const void *data, int len) OVERRIDE {
319 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
320 if (!T::SendPacket(&packet)) {
326 virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE {
327 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
328 return T::SendRtcp(&packet) ? len : -1;
336 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
337 // WebRtc Voice Engine.
338 class WebRtcVoiceMediaChannel
339 : public WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine> {
341 explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
342 virtual ~WebRtcVoiceMediaChannel();
343 virtual bool SetOptions(const AudioOptions& options);
344 virtual bool GetOptions(AudioOptions* options) const {
348 virtual bool SetRecvCodecs(const std::vector<AudioCodec> &codecs);
349 virtual bool SetSendCodecs(const std::vector<AudioCodec> &codecs);
350 virtual bool SetRecvRtpHeaderExtensions(
351 const std::vector<RtpHeaderExtension>& extensions);
352 virtual bool SetSendRtpHeaderExtensions(
353 const std::vector<RtpHeaderExtension>& extensions);
354 virtual bool SetPlayout(bool playout);
356 bool ResumePlayout();
357 virtual bool SetSend(SendFlags send);
360 virtual bool AddSendStream(const StreamParams& sp);
361 virtual bool RemoveSendStream(uint32 ssrc);
362 virtual bool AddRecvStream(const StreamParams& sp);
363 virtual bool RemoveRecvStream(uint32 ssrc);
364 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
365 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
366 virtual bool GetActiveStreams(AudioInfo::StreamList* actives);
367 virtual int GetOutputLevel();
368 virtual int GetTimeSinceLastTyping();
369 virtual void SetTypingDetectionParameters(int time_window,
370 int cost_per_typing, int reporting_threshold, int penalty_decay,
371 int type_event_delay);
372 virtual bool SetOutputScaling(uint32 ssrc, double left, double right);
373 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right);
375 virtual bool SetRingbackTone(const char *buf, int len);
376 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
377 virtual bool CanInsertDtmf();
378 virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags);
380 virtual void OnPacketReceived(rtc::Buffer* packet,
381 const rtc::PacketTime& packet_time);
382 virtual void OnRtcpReceived(rtc::Buffer* packet,
383 const rtc::PacketTime& packet_time);
384 virtual void OnReadyToSend(bool ready) {}
385 virtual bool MuteStream(uint32 ssrc, bool on);
386 virtual bool SetMaxSendBandwidth(int bps);
387 virtual bool GetStats(VoiceMediaInfo* info);
388 // Gets last reported error from WebRtc voice engine. This should be only
389 // called in response a failure.
390 virtual void GetLastMediaError(uint32* ssrc,
391 VoiceMediaChannel::Error* error);
392 bool FindSsrc(int channel_num, uint32* ssrc);
393 void OnError(uint32 ssrc, int error);
395 bool sending() const { return send_ != SEND_NOTHING; }
396 int GetReceiveChannelNum(uint32 ssrc);
397 int GetSendChannelNum(uint32 ssrc);
399 bool SetupSharedBandwidthEstimation(webrtc::VideoEngine* vie,
402 int GetLastEngineError() { return engine()->GetLastEngineError(); }
403 int GetOutputLevel(int channel);
404 bool GetRedSendCodec(const AudioCodec& red_codec,
405 const std::vector<AudioCodec>& all_codecs,
406 webrtc::CodecInst* send_codec);
407 bool EnableRtcp(int channel);
408 bool ResetRecvCodecs(int channel);
409 bool SetPlayout(int channel, bool playout);
410 static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
411 static Error WebRtcErrorToChannelError(int err_code);
414 class WebRtcVoiceChannelRenderer;
415 // Map of ssrc to WebRtcVoiceChannelRenderer object. A new object of
416 // WebRtcVoiceChannelRenderer will be created for every new stream and
417 // will be destroyed when the stream goes away.
418 typedef std::map<uint32, WebRtcVoiceChannelRenderer*> ChannelMap;
419 typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
422 void SetNack(int channel, bool nack_enabled);
423 void SetNack(const ChannelMap& channels, bool nack_enabled);
424 bool SetSendCodec(const webrtc::CodecInst& send_codec);
425 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
426 bool ChangePlayout(bool playout);
427 bool ChangeSend(SendFlags send);
428 bool ChangeSend(int channel, SendFlags send);
429 void ConfigureSendChannel(int channel);
430 bool ConfigureRecvChannel(int channel);
431 bool DeleteChannel(int channel);
432 bool InConferenceMode() const {
433 return options_.conference_mode.GetWithDefaultIfUnset(false);
435 bool IsDefaultChannel(int channel_id) const {
436 return channel_id == voe_channel();
438 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
439 bool SetSendBitrateInternal(int bps);
441 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
442 const RtpHeaderExtension* extension);
443 bool SetupSharedBweOnChannel(int voe_channel);
445 bool SetChannelRecvRtpHeaderExtensions(
447 const std::vector<RtpHeaderExtension>& extensions);
448 bool SetChannelSendRtpHeaderExtensions(
450 const std::vector<RtpHeaderExtension>& extensions);
452 rtc::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
453 std::set<int> ringback_channels_; // channels playing ringback
454 std::vector<AudioCodec> recv_codecs_;
455 std::vector<AudioCodec> send_codecs_;
456 rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
457 bool send_bitrate_setting_;
458 int send_bitrate_bps_;
459 AudioOptions options_;
461 bool desired_playout_;
464 bool typing_noise_detected_;
465 SendFlags desired_send_;
467 // shared_bwe_vie_ and shared_bwe_vie_channel_ together identifies a WebRTC
468 // VideoEngine channel that this voice channel should forward incoming packets
469 // to for Bandwidth Estimation purposes.
470 webrtc::VideoEngine* shared_bwe_vie_;
471 int shared_bwe_vie_channel_;
473 // send_channels_ contains the channels which are being used for sending.
474 // When the default channel (voe_channel) is used for sending, it is
475 // contained in send_channels_, otherwise not.
476 ChannelMap send_channels_;
477 std::vector<RtpHeaderExtension> send_extensions_;
478 uint32 default_receive_ssrc_;
479 // Note the default channel (voe_channel()) can reside in both
480 // receive_channels_ and send_channels_ in non-conference mode and in that
481 // case it will only be there if a non-zero default_receive_ssrc_ is set.
482 ChannelMap receive_channels_; // for multiple sources
483 // receive_channels_ can be read from WebRtc callback thread. Access from
484 // the WebRtc thread must be synchronized with edits on the worker thread.
485 // Reads on the worker thread are ok.
487 std::vector<RtpHeaderExtension> receive_extensions_;
488 // Do not lock this on the VoE media processor thread; potential for deadlock
490 mutable rtc::CriticalSection receive_channels_cs_;
493 } // namespace cricket
495 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_