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