/*
 *  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 PC_CHANNEL_H_
#define PC_CHANNEL_H_

#include <stddef.h>
#include <stdint.h>

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

#include "absl/types/optional.h"
#include "api/call/audio_sink.h"
#include "api/crypto/crypto_options.h"
#include "api/function_view.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "call/rtp_demuxer.h"
#include "call/rtp_packet_sink_interface.h"
#include "media/base/media_channel.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/channel_interface.h"
#include "pc/dtls_srtp_transport.h"
#include "pc/media_session.h"
#include "pc/rtp_transport.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "pc/srtp_filter.h"
#include "pc/srtp_transport.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/async_udp_socket.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/location.h"
#include "rtc_base/network.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_message.h"
#include "rtc_base/unique_id_generator.h"

namespace webrtc {
class AudioSinkInterface;
}  // namespace webrtc

namespace cricket {

struct CryptoParams;

// BaseChannel contains logic common to voice and video, including enable,
// marshaling calls to a worker and network threads, and connection and media
// monitors.
//
// BaseChannel assumes signaling and other threads are allowed to make
// synchronous calls to the worker thread, the worker thread makes synchronous
// calls only to the network thread, and the network thread can't be blocked by
// other threads.
// All methods with _n suffix must be called on network thread,
//     methods with _w suffix on worker thread
// and methods with _s suffix on signaling thread.
// Network and worker threads may be the same thread.
//
// 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 ChannelInterface,
                    // TODO(tommi): Remove has_slots inheritance.
                    public sigslot::has_slots<>,
                    // TODO(tommi): Consider implementing these interfaces
                    // via composition.
                    public MediaChannel::NetworkInterface,
                    public webrtc::RtpPacketSinkInterface {
 public:
  // If `srtp_required` is true, the channel will not send or receive any
  // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
  // The BaseChannel does not own the UniqueRandomIdGenerator so it is the
  // responsibility of the user to ensure it outlives this object.
  // TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists
  // which will make it easier to change the constructor.
  BaseChannel(rtc::Thread* worker_thread,
              rtc::Thread* network_thread,
              rtc::Thread* signaling_thread,
              std::unique_ptr<MediaChannel> media_channel,
              const std::string& content_name,
              bool srtp_required,
              webrtc::CryptoOptions crypto_options,
              rtc::UniqueRandomIdGenerator* ssrc_generator);
  virtual ~BaseChannel();
  virtual void Init_w(webrtc::RtpTransportInternal* rtp_transport);

  // Deinit may be called multiple times and is simply ignored if it's already
  // done.
  void Deinit();

  rtc::Thread* worker_thread() const { return worker_thread_; }
  rtc::Thread* network_thread() const { return network_thread_; }
  const std::string& content_name() const override { return content_name_; }
  // TODO(deadbeef): This is redundant; remove this.
  absl::string_view transport_name() const override {
    RTC_DCHECK_RUN_ON(network_thread());
    if (rtp_transport_)
      return rtp_transport_->transport_name();
    return "";
  }

  // This function returns true if using SRTP (DTLS-based keying or SDES).
  bool srtp_active() const {
    RTC_DCHECK_RUN_ON(network_thread());
    return rtp_transport_ && rtp_transport_->IsSrtpActive();
  }

  // Set an RTP level transport which could be an RtpTransport without
  // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP.
  // This can be called from any thread and it hops to the network thread
  // internally. It would replace the `SetTransports` and its variants.
  bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override;

  webrtc::RtpTransportInternal* rtp_transport() const {
    RTC_DCHECK_RUN_ON(network_thread());
    return rtp_transport_;
  }

  // Channel control
  bool SetLocalContent(const MediaContentDescription* content,
                       webrtc::SdpType type,
                       std::string* error_desc) override;
  bool SetRemoteContent(const MediaContentDescription* content,
                        webrtc::SdpType type,
                        std::string* error_desc) override;
  // Controls whether this channel will receive packets on the basis of
  // matching payload type alone. This is needed for legacy endpoints that
  // don't signal SSRCs or use MID/RID, but doesn't make sense if there is
  // more than channel of specific media type, As that creates an ambiguity.
  //
  // This method will also remove any existing streams that were bound to this
  // channel on the basis of payload type, since one of these streams might
  // actually belong to a new channel. See: crbug.com/webrtc/11477
  bool SetPayloadTypeDemuxingEnabled(bool enabled) override;

  void Enable(bool enable) override;

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

  // Used for latency measurements.
  void SetFirstPacketReceivedCallback(std::function<void()> callback) override;

  // From RtpTransport - public for testing only
  void OnTransportReadyToSend(bool ready);

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

  // RtpPacketSinkInterface overrides.
  void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;

  MediaChannel* media_channel() const override {
    return media_channel_.get();
  }

 protected:
  void set_local_content_direction(webrtc::RtpTransceiverDirection direction) {
    RTC_DCHECK_RUN_ON(worker_thread());
    local_content_direction_ = direction;
  }
  void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) {
    RTC_DCHECK_RUN_ON(worker_thread());
    remote_content_direction_ = direction;
  }
  // These methods verify that:
  // * The required content description directions have been set.
  // * The channel is enabled.
  // * And for sending:
  //   - The SRTP filter is active if it's needed.
  //   - The transport has been writable before, meaning it should be at least
  //     possible to succeed in sending a packet.
  //
  // When any of these properties change, UpdateMediaSendRecvState_w should be
  // called.
  bool IsReadyToReceiveMedia_w() const RTC_RUN_ON(worker_thread());
  bool IsReadyToSendMedia_w() const RTC_RUN_ON(worker_thread());
  rtc::Thread* signaling_thread() const { return signaling_thread_; }

  // 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 RtpTransportInternal
  void OnWritableState(bool writable);

  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);

  bool SendPacket(bool rtcp,
                  rtc::CopyOnWriteBuffer* packet,
                  const rtc::PacketOptions& options);

  void EnableMedia_w() RTC_RUN_ON(worker_thread());
  void DisableMedia_w() RTC_RUN_ON(worker_thread());

  // Performs actions if the RTP/RTCP writable state changed. This should
  // be called whenever a channel's writable state changes or when RTCP muxing
  // becomes active/inactive.
  void UpdateWritableState_n() RTC_RUN_ON(network_thread());
  void ChannelWritable_n() RTC_RUN_ON(network_thread());
  void ChannelNotWritable_n() RTC_RUN_ON(network_thread());

  bool AddRecvStream_w(const StreamParams& sp) RTC_RUN_ON(worker_thread());
  bool RemoveRecvStream_w(uint32_t ssrc) RTC_RUN_ON(worker_thread());
  void ResetUnsignaledRecvStream_w() RTC_RUN_ON(worker_thread());
  bool SetPayloadTypeDemuxingEnabled_w(bool enabled)
      RTC_RUN_ON(worker_thread());
  bool AddSendStream_w(const StreamParams& sp) RTC_RUN_ON(worker_thread());
  bool RemoveSendStream_w(uint32_t ssrc) RTC_RUN_ON(worker_thread());

  // Should be called whenever the conditions for
  // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
  // Updates the send/recv state of the media channel.
  virtual void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) = 0;

  bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
                            webrtc::SdpType type,
                            std::string* error_desc)
      RTC_RUN_ON(worker_thread());
  bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
                             webrtc::SdpType type,
                             std::string* error_desc)
      RTC_RUN_ON(worker_thread());
  virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                 webrtc::SdpType type,
                                 std::string* error_desc)
      RTC_RUN_ON(worker_thread()) = 0;
  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                  webrtc::SdpType type,
                                  std::string* error_desc)
      RTC_RUN_ON(worker_thread()) = 0;

  // Returns a list of RTP header extensions where any extension URI is unique.
  // Encrypted extensions will be either preferred or discarded, depending on
  // the current crypto_options_.
  RtpHeaderExtensions GetDeduplicatedRtpHeaderExtensions(
      const RtpHeaderExtensions& extensions);

  // Add `payload_type` to `demuxer_criteria_` if payload type demuxing is
  // enabled.
  void MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());

  void ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread());

  void UpdateRtpHeaderExtensionMap(
      const RtpHeaderExtensions& header_extensions);

  bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());

  // Return description of media channel to facilitate logging
  std::string ToString() const;

 private:
  bool ConnectToRtpTransport() RTC_RUN_ON(network_thread());
  void DisconnectFromRtpTransport() RTC_RUN_ON(network_thread());
  void SignalSentPacket_n(const rtc::SentPacket& sent_packet);

  rtc::Thread* const worker_thread_;
  rtc::Thread* const network_thread_;
  rtc::Thread* const signaling_thread_;
  rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> alive_;

  const std::string content_name_;

  std::function<void()> on_first_packet_received_
      RTC_GUARDED_BY(network_thread());

  webrtc::RtpTransportInternal* rtp_transport_
      RTC_GUARDED_BY(network_thread()) = nullptr;

  std::vector<std::pair<rtc::Socket::Option, int> > socket_options_
      RTC_GUARDED_BY(network_thread());
  std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_
      RTC_GUARDED_BY(network_thread());
  bool writable_ RTC_GUARDED_BY(network_thread()) = false;
  bool was_ever_writable_n_ RTC_GUARDED_BY(network_thread()) = false;
  bool was_ever_writable_ RTC_GUARDED_BY(worker_thread()) = false;
  const bool srtp_required_ = true;

  // TODO(tommi): This field shouldn't be necessary. It's a copy of
  // PeerConnection::GetCryptoOptions(), which is const state. It's also only
  // used to filter header extensions when calling
  // `rtp_transport_->UpdateRtpHeaderExtensionMap()` when the local/remote
  // content description is updated. Since the transport is actually owned
  // by the transport controller that also gets updated whenever the content
  // description changes, it seems we have two paths into the transports, along
  // with several thread hops via various classes (such as the Channel classes)
  // that only serve as additional layers and store duplicate state. The Jsep*
  // family of classes already apply session description updates on the network
  // thread every time it changes.
  // For the Channel classes, we should be able to get rid of:
  // * crypto_options (and fewer construction parameters)_
  // * UpdateRtpHeaderExtensionMap
  // * GetFilteredRtpHeaderExtensions
  // * Blocking thread hop to the network thread for every call to set
  //   local/remote content is updated.
  const webrtc::CryptoOptions crypto_options_;

  // MediaChannel related members that should be accessed from the worker
  // thread.
  const std::unique_ptr<MediaChannel> media_channel_;
  // Currently the `enabled_` flag is accessed from the signaling thread as
  // well, but it can be changed only when signaling thread does a synchronous
  // call to the worker thread, so it should be safe.
  bool enabled_ RTC_GUARDED_BY(worker_thread()) = false;
  bool enabled_s_ RTC_GUARDED_BY(signaling_thread()) = false;
  bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(worker_thread()) = true;
  std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread());
  std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread());
  webrtc::RtpTransceiverDirection local_content_direction_ RTC_GUARDED_BY(
      worker_thread()) = webrtc::RtpTransceiverDirection::kInactive;
  webrtc::RtpTransceiverDirection remote_content_direction_ RTC_GUARDED_BY(
      worker_thread()) = webrtc::RtpTransceiverDirection::kInactive;

  // Cached list of payload types, used if payload type demuxing is re-enabled.
  std::set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread());
  // TODO(bugs.webrtc.org/12239): Modified on worker thread, accessed
  // on network thread in RegisterRtpDemuxerSink_n (called from Init_w)
  webrtc::RtpDemuxerCriteria demuxer_criteria_;
  // Accessed on the worker thread, modified on the network thread from
  // RegisterRtpDemuxerSink_w's Invoke.
  webrtc::RtpDemuxerCriteria previous_demuxer_criteria_
      RTC_GUARDED_BY(network_thread());
  // This generator is used to generate SSRCs for local streams.
  // This is needed in cases where SSRCs are not negotiated or set explicitly
  // like in Simulcast.
  // This object is not owned by the channel so it must outlive it.
  rtc::UniqueRandomIdGenerator* const ssrc_generator_;
};

// VoiceChannel is a specialization that adds support for early media, DTMF,
// and input/output level monitoring.
class VoiceChannel : public BaseChannel {
 public:
  VoiceChannel(rtc::Thread* worker_thread,
               rtc::Thread* network_thread,
               rtc::Thread* signaling_thread,
               std::unique_ptr<VoiceMediaChannel> channel,
               const std::string& content_name,
               bool srtp_required,
               webrtc::CryptoOptions crypto_options,
               rtc::UniqueRandomIdGenerator* ssrc_generator);
  ~VoiceChannel();

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

  cricket::MediaType media_type() const override {
    return cricket::MEDIA_TYPE_AUDIO;
  }

 private:
  // overrides from BaseChannel
  void UpdateMediaSendRecvState_w() override;
  bool SetLocalContent_w(const MediaContentDescription* content,
                         webrtc::SdpType type,
                         std::string* error_desc) override;
  bool SetRemoteContent_w(const MediaContentDescription* content,
                          webrtc::SdpType type,
                          std::string* error_desc) override;

  // 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* worker_thread,
               rtc::Thread* network_thread,
               rtc::Thread* signaling_thread,
               std::unique_ptr<VideoMediaChannel> media_channel,
               const std::string& content_name,
               bool srtp_required,
               webrtc::CryptoOptions crypto_options,
               rtc::UniqueRandomIdGenerator* ssrc_generator);
  ~VideoChannel();

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

  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);

  cricket::MediaType media_type() const override {
    return cricket::MEDIA_TYPE_VIDEO;
  }

 private:
  // overrides from BaseChannel
  void UpdateMediaSendRecvState_w() override;
  bool SetLocalContent_w(const MediaContentDescription* content,
                         webrtc::SdpType type,
                         std::string* error_desc) override;
  bool SetRemoteContent_w(const MediaContentDescription* content,
                          webrtc::SdpType type,
                          std::string* error_desc) override;

  // 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_;
};

}  // namespace cricket

#endif  // PC_CHANNEL_H_