text: /*
   *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
   *
   *  Use of this source code is governed by a BSD-style license
   *  that can be found in the LICENSE file in the root of the source
   *  tree. An additional intellectual property rights grant can be found
   *  in the file PATENTS.  All contributing project authors may
   *  be found in the AUTHORS file in the root of the source tree.
   */
  
  #ifndef WEBRTC_PC_CHANNELMANAGER_H_
  #define WEBRTC_PC_CHANNELMANAGER_H_
  
  #include <memory>
  #include <string>
  #include <vector>
  
  #include "webrtc/base/fileutils.h"
  #include "webrtc/base/thread.h"
  #include "webrtc/media/base/mediaengine.h"
  #include "webrtc/pc/voicechannel.h"
  
  namespace webrtc {
  class MediaControllerInterface;
  }
  namespace cricket {
  
  class VoiceChannel;
  
  // ChannelManager allows the MediaEngine to run on a separate thread, and takes
  // care of marshalling calls between threads. It also creates and keeps track of
  // voice and video channels; by doing so, it can temporarily pause all the
  // channels when a new audio or video device is chosen. The voice and video
  // channels are stored in separate vectors, to easily allow operations on just
  // voice or just video channels.
  // ChannelManager also allows the application to discover what devices it has
  // using device manager.
  class ChannelManager {
   public:
    // For testing purposes. Allows the media engine and data media
    // engine and dev manager to be mocks.
    ChannelManager(std::unique_ptr<MediaEngineInterface> me,
                   std::unique_ptr<DataEngineInterface> dme,
                   rtc::Thread* worker_and_network);
    // Same as above, but gives an easier default DataEngine.
    ChannelManager(std::unique_ptr<MediaEngineInterface> me,
                   rtc::Thread* worker,
                   rtc::Thread* network);
    ~ChannelManager();
  
    // Accessors for the worker thread, allowing it to be set after construction,
    // but before Init. set_worker_thread will return false if called after Init.
    rtc::Thread* worker_thread() const { return worker_thread_; }
    bool set_worker_thread(rtc::Thread* thread) {
      if (initialized_) {
        return false;
      }
      worker_thread_ = thread;
      return true;
    }
    rtc::Thread* network_thread() const { return network_thread_; }
    bool set_network_thread(rtc::Thread* thread) {
      if (initialized_) {
        return false;
      }
      network_thread_ = thread;
      return true;
    }
  
    MediaEngineInterface* media_engine() { return media_engine_.get(); }
  
    // Retrieves the list of supported audio & video codec types.
    // Can be called before starting the media engine.
    void GetSupportedAudioSendCodecs(std::vector<AudioCodec>* codecs) const;
    void GetSupportedAudioReceiveCodecs(std::vector<AudioCodec>* codecs) const;
    void GetSupportedAudioRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
    void GetSupportedVideoCodecs(std::vector<VideoCodec>* codecs) const;
    void GetSupportedVideoRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
    void GetSupportedDataCodecs(std::vector<DataCodec>* codecs) const;
  
    // Indicates whether the media engine is started.
    bool initialized() const { return initialized_; }
    // Starts up the media engine.
    bool Init();
    // Shuts down the media engine.
    void Terminate();
  
    // The operations below all occur on the worker thread.
    // Creates a voice channel, to be associated with the specified session.
    VoiceChannel* CreateVoiceChannel(
        webrtc::MediaControllerInterface* media_controller,
        DtlsTransportInternal* rtp_transport,
        DtlsTransportInternal* rtcp_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required,
        const AudioOptions& options);
    // Version of the above that takes PacketTransportInternal.
    VoiceChannel* CreateVoiceChannel(
        webrtc::MediaControllerInterface* media_controller,
        rtc::PacketTransportInternal* rtp_transport,
        rtc::PacketTransportInternal* rtcp_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required,
        const AudioOptions& options);
    // Destroys a voice channel created with the Create API.
    void DestroyVoiceChannel(VoiceChannel* voice_channel);
    // Creates a video channel, synced with the specified voice channel, and
    // associated with the specified session.
    VideoChannel* CreateVideoChannel(
        webrtc::MediaControllerInterface* media_controller,
        DtlsTransportInternal* rtp_transport,
        DtlsTransportInternal* rtcp_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required,
        const VideoOptions& options);
    // Version of the above that takes PacketTransportInternal.
    VideoChannel* CreateVideoChannel(
        webrtc::MediaControllerInterface* media_controller,
        rtc::PacketTransportInternal* rtp_transport,
        rtc::PacketTransportInternal* rtcp_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required,
        const VideoOptions& options);
    // Destroys a video channel created with the Create API.
    void DestroyVideoChannel(VideoChannel* video_channel);
    RtpDataChannel* CreateRtpDataChannel(
        webrtc::MediaControllerInterface* media_controller,
        DtlsTransportInternal* rtp_transport,
        DtlsTransportInternal* rtcp_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required);
    // Destroys a data channel created with the Create API.
    void DestroyRtpDataChannel(RtpDataChannel* data_channel);
  
    // Indicates whether any channels exist.
    bool has_channels() const {
      return (!voice_channels_.empty() || !video_channels_.empty());
    }
  
    // RTX will be enabled/disabled in engines that support it. The supporting
    // engines will start offering an RTX codec. Must be called before Init().
    bool SetVideoRtxEnabled(bool enable);
  
    // Define crypto options to set on newly created channels. Doesn't change
    // options on already created channels.
    bool SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
  
    // Starts/stops the local microphone and enables polling of the input level.
    bool capturing() const { return capturing_; }
  
    // The operations below occur on the main thread.
  
    // Starts AEC dump using existing file, with a specified maximum file size in
    // bytes. When the limit is reached, logging will stop and the file will be
    // closed. If max_size_bytes is set to <= 0, no limit will be used.
    bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
  
    // Stops recording AEC dump.
    void StopAecDump();
  
   private:
    typedef std::vector<VoiceChannel*> VoiceChannels;
    typedef std::vector<VideoChannel*> VideoChannels;
    typedef std::vector<RtpDataChannel*> RtpDataChannels;
  
    void Construct(std::unique_ptr<MediaEngineInterface> me,
                   std::unique_ptr<DataEngineInterface> dme,
                   rtc::Thread* worker_thread,
                   rtc::Thread* network_thread);
    bool InitMediaEngine_w();
    void DestructorDeletes_w();
    void Terminate_w();
    bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options);
    VoiceChannel* CreateVoiceChannel_w(
        webrtc::MediaControllerInterface* media_controller,
        DtlsTransportInternal* rtp_dtls_transport,
        DtlsTransportInternal* rtcp_dtls_transport,
        rtc::PacketTransportInternal* rtp_packet_transport,
        rtc::PacketTransportInternal* rtcp_packet_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required,
        const AudioOptions& options);
    void DestroyVoiceChannel_w(VoiceChannel* voice_channel);
    VideoChannel* CreateVideoChannel_w(
        webrtc::MediaControllerInterface* media_controller,
        DtlsTransportInternal* rtp_dtls_transport,
        DtlsTransportInternal* rtcp_dtls_transport,
        rtc::PacketTransportInternal* rtp_packet_transport,
        rtc::PacketTransportInternal* rtcp_packet_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required,
        const VideoOptions& options);
    void DestroyVideoChannel_w(VideoChannel* video_channel);
    RtpDataChannel* CreateRtpDataChannel_w(
        webrtc::MediaControllerInterface* media_controller,
        DtlsTransportInternal* rtp_transport,
        DtlsTransportInternal* rtcp_transport,
        rtc::Thread* signaling_thread,
        const std::string& content_name,
        bool srtp_required);
    void DestroyRtpDataChannel_w(RtpDataChannel* data_channel);
  
    std::unique_ptr<MediaEngineInterface> media_engine_;
    std::unique_ptr<DataEngineInterface> data_media_engine_;
    bool initialized_;
    rtc::Thread* main_thread_;
    rtc::Thread* worker_thread_;
    rtc::Thread* network_thread_;
  
    VoiceChannels voice_channels_;
    VideoChannels video_channels_;
    RtpDataChannels data_channels_;
  
    bool enable_rtx_;
    rtc::CryptoOptions crypto_options_;
  
    bool capturing_;
  };
  
  }  // namespace cricket
  
  #endif  // WEBRTC_PC_CHANNELMANAGER_H_
  