321 lines
12 KiB
C
321 lines
12 KiB
C
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/*
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* libjingle
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* Copyright 2004--2011, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_SESSION_PHONE_WEBRTCVOICEENGINE_H_
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#define TALK_SESSION_PHONE_WEBRTCVOICEENGINE_H_
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#include <map>
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#include <set>
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#include <string>
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#include <vector>
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#include "talk/base/buffer.h"
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#include "talk/base/byteorder.h"
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#include "talk/base/logging.h"
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#include "talk/base/scoped_ptr.h"
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#include "talk/base/stream.h"
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#include "talk/session/phone/channel.h"
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#include "talk/session/phone/mediaengine.h"
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#include "talk/session/phone/rtputils.h"
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#include "talk/session/phone/webrtccommon.h"
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namespace cricket {
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// WebRtcSoundclipStream is an adapter object that allows a memory stream to be
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// passed into WebRtc, and support looping.
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class WebRtcSoundclipStream : public webrtc::InStream {
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public:
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WebRtcSoundclipStream(const char* buf, size_t len)
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: mem_(buf, len), loop_(true) {
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}
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void set_loop(bool loop) { loop_ = loop; }
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virtual int Read(void* buf, int len);
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virtual int Rewind();
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private:
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talk_base::MemoryStream mem_;
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bool loop_;
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};
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// WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
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// For now we just dump the data.
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class WebRtcMonitorStream : public webrtc::OutStream {
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virtual bool Write(const void *buf, int len) {
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return true;
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}
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};
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class AudioDeviceModule;
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class VoETraceWrapper;
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class VoEWrapper;
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class WebRtcSoundclipMedia;
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class WebRtcVoiceMediaChannel;
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// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
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// It uses the WebRtc VoiceEngine library for audio handling.
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class WebRtcVoiceEngine
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: public webrtc::VoiceEngineObserver,
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public webrtc::TraceCallback {
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public:
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WebRtcVoiceEngine();
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WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
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webrtc::AudioDeviceModule* adm_sc);
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// Dependency injection for testing.
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WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
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VoEWrapper* voe_wrapper_sc,
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VoETraceWrapper* tracing);
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~WebRtcVoiceEngine();
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bool Init();
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void Terminate();
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int GetCapabilities();
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VoiceMediaChannel* CreateChannel();
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SoundclipMedia* CreateSoundclip();
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bool SetOptions(int options);
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bool SetDevices(const Device* in_device, const Device* out_device);
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bool GetOutputVolume(int* level);
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bool SetOutputVolume(int level);
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int GetInputLevel();
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bool SetLocalMonitor(bool enable);
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const std::vector<AudioCodec>& codecs();
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bool FindCodec(const AudioCodec& codec);
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bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
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void SetLogging(int min_sev, const char* filter);
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// For tracking WebRtc channels. Needed because we have to pause them
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// all when switching devices.
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// May only be called by WebRtcVoiceMediaChannel.
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void RegisterChannel(WebRtcVoiceMediaChannel *channel);
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void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
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// May only be called by WebRtcSoundclipMedia.
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void RegisterSoundclip(WebRtcSoundclipMedia *channel);
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void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
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// Called by WebRtcVoiceMediaChannel to set a gain offset from
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// the default AGC target level.
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bool AdjustAgcLevel(int delta);
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// Called by WebRtcVoiceMediaChannel to configure echo cancellation
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// and noise suppression modes.
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bool SetConferenceMode(bool enable);
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VoEWrapper* voe() { return voe_wrapper_.get(); }
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VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
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int GetLastEngineError();
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private:
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typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
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typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
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struct CodecPref {
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const char* name;
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int clockrate;
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};
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void Construct();
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bool InitInternal();
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void ApplyLogging();
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virtual void Print(const webrtc::TraceLevel level,
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const char* trace_string, const int length);
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virtual void CallbackOnError(const int channel, const int errCode);
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static int GetCodecPreference(const char *name, int clockrate);
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// Given the device type, name, and id, find device id. Return true and
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// set the output parameter rtc_id if successful.
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bool FindWebRtcAudioDeviceId(
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bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
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bool FindChannelAndSsrc(int channel_num,
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WebRtcVoiceMediaChannel** channel,
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uint32* ssrc) const;
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bool ChangeLocalMonitor(bool enable);
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bool PauseLocalMonitor();
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bool ResumeLocalMonitor();
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static const int kDefaultLogSeverity = talk_base::LS_WARNING;
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static const CodecPref kCodecPrefs[];
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// The primary instance of WebRtc VoiceEngine.
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talk_base::scoped_ptr<VoEWrapper> voe_wrapper_;
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// A secondary instance, for playing out soundclips (on the 'ring' device).
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talk_base::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
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talk_base::scoped_ptr<VoETraceWrapper> tracing_;
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// The external audio device manager
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webrtc::AudioDeviceModule* adm_;
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webrtc::AudioDeviceModule* adm_sc_;
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int log_level_;
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bool is_dumping_aec_;
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std::vector<AudioCodec> codecs_;
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bool desired_local_monitor_enable_;
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talk_base::scoped_ptr<WebRtcMonitorStream> monitor_;
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SoundclipList soundclips_;
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ChannelList channels_;
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// channels_ can be read from WebRtc callback thread. We need a lock on that
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// callback as well as the RegisterChannel/UnregisterChannel.
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talk_base::CriticalSection channels_cs_;
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webrtc::AgcConfig default_agc_config_;
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};
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// WebRtcMediaChannel is a class that implements the common WebRtc channel
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// functionality.
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template <class T, class E>
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class WebRtcMediaChannel : public T, public webrtc::Transport {
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public:
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WebRtcMediaChannel(E *engine, int channel)
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: engine_(engine), voe_channel_(channel), sequence_number_(-1) {}
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E *engine() { return engine_; }
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int voe_channel() const { return voe_channel_; }
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bool valid() const { return voe_channel_ != -1; }
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protected:
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// implements Transport interface
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virtual int SendPacket(int channel, const void *data, int len) {
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if (!T::network_interface_) {
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return -1;
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}
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// We need to store the sequence number to be able to pick up
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// the same sequence when the device is restarted.
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// TODO(oja): Remove when WebRtc has fixed the problem.
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int seq_num;
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if (!GetRtpSeqNum(data, len, &seq_num)) {
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return -1;
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}
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if (sequence_number() == -1) {
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LOG(INFO) << "WebRtcVoiceMediaChannel sends first packet seqnum="
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<< seq_num;
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}
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sequence_number_ = seq_num;
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talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
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return T::network_interface_->SendPacket(&packet) ? len : -1;
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}
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virtual int SendRTCPPacket(int channel, const void *data, int len) {
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if (!T::network_interface_) {
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return -1;
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}
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talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
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return T::network_interface_->SendRtcp(&packet) ? len : -1;
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}
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int sequence_number() const {
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return sequence_number_;
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}
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private:
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E *engine_;
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int voe_channel_;
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int sequence_number_;
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};
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// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
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// WebRtc Voice Engine.
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class WebRtcVoiceMediaChannel
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: public WebRtcMediaChannel<VoiceMediaChannel,
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WebRtcVoiceEngine> {
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public:
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explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
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virtual ~WebRtcVoiceMediaChannel();
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virtual bool SetOptions(int options);
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virtual bool SetRecvCodecs(const std::vector<AudioCodec> &codecs);
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virtual bool SetSendCodecs(const std::vector<AudioCodec> &codecs);
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virtual bool SetRecvRtpHeaderExtensions(
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const std::vector<RtpHeaderExtension>& extensions);
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virtual bool SetSendRtpHeaderExtensions(
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const std::vector<RtpHeaderExtension>& extensions);
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virtual bool SetPlayout(bool playout);
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bool PausePlayout();
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bool ResumePlayout();
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virtual bool SetSend(SendFlags send);
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bool PauseSend();
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bool ResumeSend();
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virtual bool AddStream(uint32 ssrc);
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virtual bool RemoveStream(uint32 ssrc);
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virtual bool GetActiveStreams(AudioInfo::StreamList* actives);
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virtual int GetOutputLevel();
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virtual bool SetRingbackTone(const char *buf, int len);
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virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
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virtual bool PressDTMF(int event, bool playout);
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virtual void OnPacketReceived(talk_base::Buffer* packet);
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virtual void OnRtcpReceived(talk_base::Buffer* packet);
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virtual void SetSendSsrc(uint32 id);
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virtual bool SetRtcpCName(const std::string& cname);
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virtual bool Mute(bool mute);
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virtual bool SetSendBandwidth(bool autobw, int bps) { return false; }
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virtual bool GetStats(VoiceMediaInfo* info);
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// Gets last reported error from WebRtc voice engine. This should be only
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// called in response a failure.
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virtual void GetLastMediaError(uint32* ssrc,
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VoiceMediaChannel::Error* error);
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bool FindSsrc(int channel_num, uint32* ssrc);
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void OnError(uint32 ssrc, int error);
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protected:
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int GetLastEngineError() { return engine()->GetLastEngineError(); }
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int GetChannel(uint32 ssrc);
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int GetOutputLevel(int channel);
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bool GetRedSendCodec(const AudioCodec& red_codec,
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const std::vector<AudioCodec>& all_codecs,
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webrtc::CodecInst* send_codec);
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bool EnableRtcp(int channel);
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bool SetPlayout(int channel, bool playout);
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static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
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static Error WebRtcErrorToChannelError(int err_code);
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private:
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// Tandberg-bridged conferences require a -10dB gain adjustment,
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// which is actually +10 in AgcConfig.targetLeveldBOv
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static const int kTandbergDbAdjustment = 10;
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bool ChangePlayout(bool playout);
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bool ChangeSend(SendFlags send);
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typedef std::map<uint32, int> ChannelMap;
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talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
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std::set<int> ringback_channels_; // channels playing ringback
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int channel_options_;
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bool agc_adjusted_;
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bool dtmf_allowed_;
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bool desired_playout_;
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bool playout_;
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SendFlags desired_send_;
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SendFlags send_;
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ChannelMap mux_channels_; // for multiple sources
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// mux_channels_ can be read from WebRtc callback thread. Accesses off the
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// WebRtc thread must be synchronized with edits on the worker thread. Reads
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// on the worker thread are ok.
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mutable talk_base::CriticalSection mux_channels_cs_;
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};
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}
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#endif // TALK_SESSION_PHONE_WEBRTCVOICEENGINE_H_
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