/*
 *  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_CHANNEL_H_
#define WEBRTC_PC_CHANNEL_H_

#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>

#include "webrtc/audio_sink.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/network.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/window.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/base/videocapturer.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/p2p/base/transportcontroller.h"
#include "webrtc/p2p/client/socketmonitor.h"
#include "webrtc/pc/audiomonitor.h"
#include "webrtc/pc/bundlefilter.h"
#include "webrtc/pc/mediamonitor.h"
#include "webrtc/pc/mediasession.h"
#include "webrtc/pc/rtcpmuxfilter.h"
#include "webrtc/pc/srtpfilter.h"

namespace webrtc {
class AudioSinkInterface;
}  // namespace webrtc

namespace cricket {

struct CryptoParams;
class MediaContentDescription;

enum SinkType {
  SINK_PRE_CRYPTO,  // Sink packets before encryption or after decryption.
  SINK_POST_CRYPTO  // Sink packets after encryption or before decryption.
};

// BaseChannel contains logic common to voice and video, including
// enable, marshaling calls to a worker thread, and
// connection and media monitors.
//
// WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
// This is required to avoid a data race between the destructor modifying the
// vtable, and the media channel's thread using BaseChannel as the
// NetworkInterface.

class BaseChannel
    : public rtc::MessageHandler, public sigslot::has_slots<>,
      public MediaChannel::NetworkInterface,
      public ConnectionStatsGetter {
 public:
  BaseChannel(rtc::Thread* thread,
              MediaChannel* channel,
              TransportController* transport_controller,
              const std::string& content_name,
              bool rtcp);
  virtual ~BaseChannel();
  bool Init();
  // Deinit may be called multiple times and is simply ignored if it's alreay
  // done.
  void Deinit();

  rtc::Thread* worker_thread() const { return worker_thread_; }
  const std::string& content_name() const { return content_name_; }
  const std::string& transport_name() const { return transport_name_; }
  TransportChannel* transport_channel() const {
    return transport_channel_;
  }
  TransportChannel* rtcp_transport_channel() const {
    return rtcp_transport_channel_;
  }
  bool enabled() const { return enabled_; }

  // This function returns true if we are using SRTP.
  bool secure() const { return srtp_filter_.IsActive(); }
  // The following function returns true if we are using
  // DTLS-based keying. If you turned off SRTP later, however
  // you could have secure() == false and dtls_secure() == true.
  bool secure_dtls() const { return dtls_keyed_; }
  // This function returns true if we require secure channel for call setup.
  bool secure_required() const { return secure_required_; }

  bool writable() const { return writable_; }

  // Activate RTCP mux, regardless of the state so far.  Once
  // activated, it can not be deactivated, and if the remote
  // description doesn't support RTCP mux, setting the remote
  // description will fail.
  void ActivateRtcpMux();
  bool SetTransport(const std::string& transport_name);
  bool PushdownLocalDescription(const SessionDescription* local_desc,
                                ContentAction action,
                                std::string* error_desc);
  bool PushdownRemoteDescription(const SessionDescription* remote_desc,
                                ContentAction action,
                                std::string* error_desc);
  // Channel control
  bool SetLocalContent(const MediaContentDescription* content,
                       ContentAction action,
                       std::string* error_desc);
  bool SetRemoteContent(const MediaContentDescription* content,
                        ContentAction action,
                        std::string* error_desc);

  bool Enable(bool enable);

  // Multiplexing
  bool AddRecvStream(const StreamParams& sp);
  bool RemoveRecvStream(uint32_t ssrc);
  bool AddSendStream(const StreamParams& sp);
  bool RemoveSendStream(uint32_t ssrc);

  // Monitoring
  void StartConnectionMonitor(int cms);
  void StopConnectionMonitor();
  // For ConnectionStatsGetter, used by ConnectionMonitor
  bool GetConnectionStats(ConnectionInfos* infos) override;

  BundleFilter* bundle_filter() { return &bundle_filter_; }

  const std::vector<StreamParams>& local_streams() const {
    return local_streams_;
  }
  const std::vector<StreamParams>& remote_streams() const {
    return remote_streams_;
  }

  sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure;
  void SignalDtlsSetupFailure_w(bool rtcp);
  void SignalDtlsSetupFailure_s(bool rtcp);

  // Used for latency measurements.
  sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;

  // Made public for easier testing.
  void SetReadyToSend(bool rtcp, bool ready);

  // Only public for unit tests.  Otherwise, consider protected.
  int SetOption(SocketType type, rtc::Socket::Option o, int val)
      override;

  SrtpFilter* srtp_filter() { return &srtp_filter_; }

 protected:
  virtual MediaChannel* media_channel() const { return media_channel_; }
  // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
  // true). Gets the transport channels from |transport_controller_|.
  bool SetTransport_w(const std::string& transport_name);

  void set_transport_channel(TransportChannel* transport);
  void set_rtcp_transport_channel(TransportChannel* transport,
                                  bool update_writablity);

  bool was_ever_writable() const { return was_ever_writable_; }
  void set_local_content_direction(MediaContentDirection direction) {
    local_content_direction_ = direction;
  }
  void set_remote_content_direction(MediaContentDirection direction) {
    remote_content_direction_ = direction;
  }
  void set_secure_required(bool secure_required) {
    secure_required_ = secure_required;
  }
  bool IsReadyToReceive() const;
  bool IsReadyToSend() const;
  rtc::Thread* signaling_thread() {
    return transport_controller_->signaling_thread();
  }
  bool rtcp_transport_enabled() const { return rtcp_transport_enabled_; }

  void ConnectToTransportChannel(TransportChannel* tc);
  void DisconnectFromTransportChannel(TransportChannel* tc);

  void FlushRtcpMessages();

  // NetworkInterface implementation, called by MediaEngine
  bool SendPacket(rtc::CopyOnWriteBuffer* packet,
                  const rtc::PacketOptions& options) override;
  bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
                const rtc::PacketOptions& options) override;

  // From TransportChannel
  void OnWritableState(TransportChannel* channel);
  virtual void OnChannelRead(TransportChannel* channel,
                             const char* data,
                             size_t len,
                             const rtc::PacketTime& packet_time,
                             int flags);
  void OnReadyToSend(TransportChannel* channel);

  void OnDtlsState(TransportChannel* channel, DtlsTransportState state);

  void OnSelectedCandidatePairChanged(
      TransportChannel* channel,
      CandidatePairInterface* selected_candidate_pair,
      int last_sent_packet_id);

  bool PacketIsRtcp(const TransportChannel* channel, const char* data,
                    size_t len);
  bool SendPacket(bool rtcp,
                  rtc::CopyOnWriteBuffer* packet,
                  const rtc::PacketOptions& options);
  virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
  void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
                    const rtc::PacketTime& packet_time);

  void EnableMedia_w();
  void DisableMedia_w();
  void UpdateWritableState_w();
  void ChannelWritable_w();
  void ChannelNotWritable_w();
  bool AddRecvStream_w(const StreamParams& sp);
  bool RemoveRecvStream_w(uint32_t ssrc);
  bool AddSendStream_w(const StreamParams& sp);
  bool RemoveSendStream_w(uint32_t ssrc);
  virtual bool ShouldSetupDtlsSrtp() const;
  // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
  // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
  bool SetupDtlsSrtp(bool rtcp_channel);
  void MaybeSetupDtlsSrtp_w();
  // Set the DTLS-SRTP cipher policy on this channel as appropriate.
  bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp);

  virtual void ChangeState() = 0;

  // Gets the content info appropriate to the channel (audio or video).
  virtual const ContentInfo* GetFirstContent(
      const SessionDescription* sdesc) = 0;
  bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
                            ContentAction action,
                            std::string* error_desc);
  bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
                             ContentAction action,
                             std::string* error_desc);
  virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                 ContentAction action,
                                 std::string* error_desc) = 0;
  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc) = 0;
  bool SetRtpTransportParameters_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   ContentSource src,
                                   std::string* error_desc);

  // Helper method to get RTP Absoulute SendTime extension header id if
  // present in remote supported extensions list.
  void MaybeCacheRtpAbsSendTimeHeaderExtension(
      const std::vector<RtpHeaderExtension>& extensions);

  bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
                       bool* dtls,
                       std::string* error_desc);
  bool SetSrtp_w(const std::vector<CryptoParams>& params,
                 ContentAction action,
                 ContentSource src,
                 std::string* error_desc);
  void ActivateRtcpMux_w();
  bool SetRtcpMux_w(bool enable,
                    ContentAction action,
                    ContentSource src,
                    std::string* error_desc);

  // From MessageHandler
  void OnMessage(rtc::Message* pmsg) override;

  // Handled in derived classes
  // Get the SRTP crypto suites to use for RTP media
  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
  virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
      const std::vector<ConnectionInfo>& infos) = 0;

  // Helper function for invoking bool-returning methods on the worker thread.
  template <class FunctorT>
  bool InvokeOnWorker(const FunctorT& functor) {
    return worker_thread_->Invoke<bool>(functor);
  }

 private:
  rtc::Thread* worker_thread_;
  TransportController* transport_controller_;
  MediaChannel* media_channel_;
  std::vector<StreamParams> local_streams_;
  std::vector<StreamParams> remote_streams_;

  const std::string content_name_;
  std::string transport_name_;
  bool rtcp_transport_enabled_;
  TransportChannel* transport_channel_;
  std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
  TransportChannel* rtcp_transport_channel_;
  std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
  SrtpFilter srtp_filter_;
  RtcpMuxFilter rtcp_mux_filter_;
  BundleFilter bundle_filter_;
  std::unique_ptr<ConnectionMonitor> connection_monitor_;
  bool enabled_;
  bool writable_;
  bool rtp_ready_to_send_;
  bool rtcp_ready_to_send_;
  bool was_ever_writable_;
  MediaContentDirection local_content_direction_;
  MediaContentDirection remote_content_direction_;
  bool has_received_packet_;
  bool dtls_keyed_;
  bool secure_required_;
  int rtp_abs_sendtime_extn_id_;
};

// VoiceChannel is a specialization that adds support for early media, DTMF,
// and input/output level monitoring.
class VoiceChannel : public BaseChannel {
 public:
  VoiceChannel(rtc::Thread* thread,
               MediaEngineInterface* media_engine,
               VoiceMediaChannel* channel,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
  ~VoiceChannel();
  bool Init();

  // Configure sending media on the stream with SSRC |ssrc|
  // If there is only one sending stream SSRC 0 can be used.
  bool SetAudioSend(uint32_t ssrc,
                    bool enable,
                    const AudioOptions* options,
                    AudioSource* source);

  // downcasts a MediaChannel
  virtual VoiceMediaChannel* media_channel() const {
    return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
  }

  void SetEarlyMedia(bool enable);
  // This signal is emitted when we have gone a period of time without
  // receiving early media. When received, a UI should start playing its
  // own ringing sound
  sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;

  // Returns if the telephone-event has been negotiated.
  bool CanInsertDtmf();
  // Send and/or play a DTMF |event| according to the |flags|.
  // The DTMF out-of-band signal will be used on sending.
  // The |ssrc| should be either 0 or a valid send stream ssrc.
  // The valid value for the |event| are 0 which corresponding to DTMF
  // event 0-9, *, #, A-D.
  bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
  bool SetOutputVolume(uint32_t ssrc, double volume);
  void SetRawAudioSink(uint32_t ssrc,
                       std::unique_ptr<webrtc::AudioSinkInterface> sink);
  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
  bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);

  // Get statistics about the current media session.
  bool GetStats(VoiceMediaInfo* stats);

  // Monitoring functions
  sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
      SignalConnectionMonitor;

  void StartMediaMonitor(int cms);
  void StopMediaMonitor();
  sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;

  void StartAudioMonitor(int cms);
  void StopAudioMonitor();
  bool IsAudioMonitorRunning() const;
  sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;

  int GetInputLevel_w();
  int GetOutputLevel_w();
  void GetActiveStreams_w(AudioInfo::StreamList* actives);
  webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);

 private:
  // overrides from BaseChannel
  virtual void OnChannelRead(TransportChannel* channel,
                             const char* data, size_t len,
                             const rtc::PacketTime& packet_time,
                             int flags);
  virtual void ChangeState();
  virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
  virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                 ContentAction action,
                                 std::string* error_desc);
  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
  void HandleEarlyMediaTimeout();
  bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
  bool SetOutputVolume_w(uint32_t ssrc, double volume);
  bool GetStats_w(VoiceMediaInfo* stats);

  virtual void OnMessage(rtc::Message* pmsg);
  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
  virtual void OnConnectionMonitorUpdate(
      ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
  virtual void OnMediaMonitorUpdate(
      VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
  void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);

  static const int kEarlyMediaTimeout = 1000;
  MediaEngineInterface* media_engine_;
  bool received_media_;
  std::unique_ptr<VoiceMediaMonitor> media_monitor_;
  std::unique_ptr<AudioMonitor> audio_monitor_;

  // Last AudioSendParameters sent down to the media_channel() via
  // SetSendParameters.
  AudioSendParameters last_send_params_;
  // Last AudioRecvParameters sent down to the media_channel() via
  // SetRecvParameters.
  AudioRecvParameters last_recv_params_;
};

// VideoChannel is a specialization for video.
class VideoChannel : public BaseChannel {
 public:
  VideoChannel(rtc::Thread* thread,
               VideoMediaChannel* channel,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
  ~VideoChannel();
  bool Init();

  // downcasts a MediaChannel
  virtual VideoMediaChannel* media_channel() const {
    return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
  }

  bool SetSink(uint32_t ssrc, rtc::VideoSinkInterface<VideoFrame>* sink);
  bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer);
  // Get statistics about the current media session.
  bool GetStats(VideoMediaInfo* stats);

  sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
      SignalConnectionMonitor;

  void StartMediaMonitor(int cms);
  void StopMediaMonitor();
  sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;

  bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
  bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);

 private:
  // overrides from BaseChannel
  virtual void ChangeState();
  virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
  virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                 ContentAction action,
                                 std::string* error_desc);
  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
  bool GetStats_w(VideoMediaInfo* stats);
  webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);

  virtual void OnMessage(rtc::Message* pmsg);
  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
  virtual void OnConnectionMonitorUpdate(
      ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
  virtual void OnMediaMonitorUpdate(
      VideoMediaChannel* media_channel, const VideoMediaInfo& info);

  std::unique_ptr<VideoMediaMonitor> media_monitor_;

  // Last VideoSendParameters sent down to the media_channel() via
  // SetSendParameters.
  VideoSendParameters last_send_params_;
  // Last VideoRecvParameters sent down to the media_channel() via
  // SetRecvParameters.
  VideoRecvParameters last_recv_params_;
};

// DataChannel is a specialization for data.
class DataChannel : public BaseChannel {
 public:
  DataChannel(rtc::Thread* thread,
              DataMediaChannel* media_channel,
              TransportController* transport_controller,
              const std::string& content_name,
              bool rtcp);
  ~DataChannel();
  bool Init();

  virtual bool SendData(const SendDataParams& params,
                        const rtc::CopyOnWriteBuffer& payload,
                        SendDataResult* result);

  void StartMediaMonitor(int cms);
  void StopMediaMonitor();

  // Should be called on the signaling thread only.
  bool ready_to_send_data() const {
    return ready_to_send_data_;
  }

  sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
  sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
      SignalConnectionMonitor;
  sigslot::signal3<DataChannel*, const ReceiveDataParams&,
      const rtc::CopyOnWriteBuffer&> SignalDataReceived;
  // Signal for notifying when the channel becomes ready to send data.
  // That occurs when the channel is enabled, the transport is writable,
  // both local and remote descriptions are set, and the channel is unblocked.
  sigslot::signal1<bool> SignalReadyToSendData;
  // Signal for notifying that the remote side has closed the DataChannel.
  sigslot::signal1<uint32_t> SignalStreamClosedRemotely;

 protected:
  // downcasts a MediaChannel.
  virtual DataMediaChannel* media_channel() const {
    return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
  }

 private:
  struct SendDataMessageData : public rtc::MessageData {
    SendDataMessageData(const SendDataParams& params,
                        const rtc::CopyOnWriteBuffer* payload,
                        SendDataResult* result)
        : params(params),
          payload(payload),
          result(result),
          succeeded(false) {
    }

    const SendDataParams& params;
    const rtc::CopyOnWriteBuffer* payload;
    SendDataResult* result;
    bool succeeded;
  };

  struct DataReceivedMessageData : public rtc::MessageData {
    // We copy the data because the data will become invalid after we
    // handle DataMediaChannel::SignalDataReceived but before we fire
    // SignalDataReceived.
    DataReceivedMessageData(
        const ReceiveDataParams& params, const char* data, size_t len)
        : params(params),
          payload(data, len) {
    }
    const ReceiveDataParams params;
    const rtc::CopyOnWriteBuffer payload;
  };

  typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;

  // overrides from BaseChannel
  virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
  // If data_channel_type_ is DCT_NONE, set it.  Otherwise, check that
  // it's the same as what was set previously.  Returns false if it's
  // set to one type one type and changed to another type later.
  bool SetDataChannelType(DataChannelType new_data_channel_type,
                          std::string* error_desc);
  // Same as SetDataChannelType, but extracts the type from the
  // DataContentDescription.
  bool SetDataChannelTypeFromContent(const DataContentDescription* content,
                                     std::string* error_desc);
  virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                 ContentAction action,
                                 std::string* error_desc);
  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
  virtual void ChangeState();
  virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);

  virtual void OnMessage(rtc::Message* pmsg);
  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
  virtual void OnConnectionMonitorUpdate(
      ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
  virtual void OnMediaMonitorUpdate(
      DataMediaChannel* media_channel, const DataMediaInfo& info);
  virtual bool ShouldSetupDtlsSrtp() const;
  void OnDataReceived(
      const ReceiveDataParams& params, const char* data, size_t len);
  void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
  void OnDataChannelReadyToSend(bool writable);
  void OnStreamClosedRemotely(uint32_t sid);

  std::unique_ptr<DataMediaMonitor> media_monitor_;
  // TODO(pthatcher): Make a separate SctpDataChannel and
  // RtpDataChannel instead of using this.
  DataChannelType data_channel_type_;
  bool ready_to_send_data_;

  // Last DataSendParameters sent down to the media_channel() via
  // SetSendParameters.
  DataSendParameters last_send_params_;
  // Last DataRecvParameters sent down to the media_channel() via
  // SetRecvParameters.
  DataRecvParameters last_recv_params_;
};

}  // namespace cricket

#endif  // WEBRTC_PC_CHANNEL_H_