path: root/media/webrtc/signaling/src/media-conduit/VideoConduit.h

/* This Source Code Form is subject to the terms of the Mozilla Public
 * License, v. 2.0. If a copy of the MPL was not distributed with this file,
 * You can obtain one at http://mozilla.org/MPL/2.0/. */

#ifndef VIDEO_SESSION_H_
#define VIDEO_SESSION_H_

#include "nsAutoPtr.h"
#include "mozilla/Attributes.h"
#include "mozilla/Atomics.h"

#include "MediaConduitInterface.h"
#include "MediaEngineWrapper.h"
#include "CodecStatistics.h"
#include "LoadManagerFactory.h"
#include "LoadManager.h"
#include "runnable_utils.h"

// conflicts with #include of scoped_ptr.h
#undef FF
// Video Engine Includes
#include "webrtc/common_types.h"
#ifdef FF
#undef FF // Avoid name collision between scoped_ptr.h and nsCRTGlue.h.
#endif
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/video_engine/include/vie_base.h"
#include "webrtc/video_engine/include/vie_capture.h"
#include "webrtc/video_engine/include/vie_codec.h"
#include "webrtc/video_engine/include/vie_external_codec.h"
#include "webrtc/video_engine/include/vie_render.h"
#include "webrtc/video_engine/include/vie_network.h"
#include "webrtc/video_engine/include/vie_rtp_rtcp.h"

/** This file hosts several structures identifying different aspects
 * of a RTP Session.
 */

 using  webrtc::ViEBase;
 using  webrtc::ViENetwork;
 using  webrtc::ViECodec;
 using  webrtc::ViECapture;
 using  webrtc::ViERender;
 using  webrtc::ViEExternalCapture;
 using  webrtc::ViEExternalCodec;

namespace mozilla {

class WebrtcAudioConduit;
class nsThread;

// Interface of external video encoder for WebRTC.
class WebrtcVideoEncoder:public VideoEncoder
                         ,public webrtc::VideoEncoder
{};

// Interface of external video decoder for WebRTC.
class WebrtcVideoDecoder:public VideoDecoder
                         ,public webrtc::VideoDecoder
{};

/**
 * Concrete class for Video session. Hooks up
 *  - media-source and target to external transport
 */
class WebrtcVideoConduit : public VideoSessionConduit
                         , public webrtc::Transport
                         , public webrtc::ExternalRenderer
{
public:
  //VoiceEngine defined constant for Payload Name Size.
  static const unsigned int CODEC_PLNAME_SIZE;

  /**
   * Set up A/V sync between this (incoming) VideoConduit and an audio conduit.
   */
  void SyncTo(WebrtcAudioConduit *aConduit);

  /**
   * Function to attach Renderer end-point for the Media-Video conduit.
   * @param aRenderer : Reference to the concrete Video renderer implementation
   * Note: Multiple invocations of this API shall remove an existing renderer
   * and attaches the new to the Conduit.
   */
  virtual MediaConduitErrorCode AttachRenderer(RefPtr<VideoRenderer> aVideoRenderer) override;
  virtual void DetachRenderer() override;

  /**
   * APIs used by the registered external transport to this Conduit to
   * feed in received RTP Frames to the VideoEngine for decoding
   */
  virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len) override;

  /**
   * APIs used by the registered external transport to this Conduit to
   * feed in received RTP Frames to the VideoEngine for decoding
   */
  virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len) override;

  virtual MediaConduitErrorCode StopTransmitting() override;
  virtual MediaConduitErrorCode StartTransmitting() override;
  virtual MediaConduitErrorCode StopReceiving() override;
  virtual MediaConduitErrorCode StartReceiving() override;

  /**
   * Function to configure sending codec mode for different content
   */
  virtual MediaConduitErrorCode ConfigureCodecMode(webrtc::VideoCodecMode) override;

   /**
   * Function to configure send codec for the video session
   * @param sendSessionConfig: CodecConfiguration
   * @result: On Success, the video engine is configured with passed in codec for send
   *          On failure, video engine transmit functionality is disabled.
   * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
   *        transmission sub-system on the engine.
   */
  virtual MediaConduitErrorCode ConfigureSendMediaCodec(const VideoCodecConfig* codecInfo) override;

  /**
   * Function to configure list of receive codecs for the video session
   * @param sendSessionConfig: CodecConfiguration
   * @result: On Success, the video engine is configured with passed in codec for send
   *          Also the playout is enabled.
   *          On failure, video engine transmit functionality is disabled.
   * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
   *        transmission sub-system on the engine.
   */
   virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
                               const std::vector<VideoCodecConfig* >& codecConfigList) override;

  /**
   * Register Transport for this Conduit. RTP and RTCP frames from the VideoEngine
   * shall be passed to the registered transport for transporting externally.
   */
  virtual MediaConduitErrorCode SetTransmitterTransport(RefPtr<TransportInterface> aTransport) override;

  virtual MediaConduitErrorCode SetReceiverTransport(RefPtr<TransportInterface> aTransport) override;

  /**
   * Function to set the encoding bitrate limits based on incoming frame size and rate
   * @param width, height: dimensions of the frame
   * @param cap: user-enforced max bitrate, or 0
   * @param aLastFramerateTenths: holds the current input framerate
   * @param out_start, out_min, out_max: bitrate results
   */
  void SelectBitrates(unsigned short width,
                      unsigned short height,
                      unsigned int cap,
                      mozilla::Atomic<int32_t, mozilla::Relaxed>& aLastFramerateTenths,
                      unsigned int& out_min,
                      unsigned int& out_start,
                      unsigned int& out_max);

  /**
   * Function to select and change the encoding resolution based on incoming frame size
   * and current available bandwidth.
   * @param width, height: dimensions of the frame
   * @param frame: optional frame to submit for encoding after reconfig
   */
  bool SelectSendResolution(unsigned short width,
                            unsigned short height,
                            webrtc::I420VideoFrame *frame);

  /**
   * Function to reconfigure the current send codec for a different
   * width/height/framerate/etc.
   * @param width, height: dimensions of the frame
   * @param frame: optional frame to submit for encoding after reconfig
   */
  nsresult ReconfigureSendCodec(unsigned short width,
                                unsigned short height,
                                webrtc::I420VideoFrame *frame);

  /**
   * Function to select and change the encoding frame rate based on incoming frame rate
   * and max-mbps setting.
   * @param current framerate
   * @result new framerate
   */
  unsigned int SelectSendFrameRate(unsigned int framerate) const;

  /**
   * Function to deliver a capture video frame for encoding and transport
   * @param video_frame: pointer to captured video-frame.
   * @param video_frame_length: size of the frame
   * @param width, height: dimensions of the frame
   * @param video_type: Type of the video frame - I420, RAW
   * @param captured_time: timestamp when the frame was captured.
   *                       if 0 timestamp is automatcally generated by the engine.
   *NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked
   *       This ensures the inserted video-frames can be transmitted by the conduit
   */
  virtual MediaConduitErrorCode SendVideoFrame(unsigned char* video_frame,
                                                unsigned int video_frame_length,
                                                unsigned short width,
                                                unsigned short height,
                                                VideoType video_type,
                                                uint64_t capture_time) override;
  virtual MediaConduitErrorCode SendVideoFrame(webrtc::I420VideoFrame& frame) override;

  /**
   * Set an external encoder object |encoder| to the payload type |pltype|
   * for sender side codec.
   */
  virtual MediaConduitErrorCode SetExternalSendCodec(VideoCodecConfig* config,
                                                     VideoEncoder* encoder) override;

  /**
   * Set an external decoder object |decoder| to the payload type |pltype|
   * for receiver side codec.
   */
  virtual MediaConduitErrorCode SetExternalRecvCodec(VideoCodecConfig* config,
                                                     VideoDecoder* decoder) override;

  /**
  * Enables use of Rtp Stream Id, and sets the extension ID.
  */
  virtual MediaConduitErrorCode EnableRTPStreamIdExtension(bool enabled, uint8_t id) override;

  /**
   * Webrtc transport implementation to send and receive RTP packet.
   * VideoConduit registers itself as ExternalTransport to the VideoEngine
   */
  virtual int SendPacket(int channel, const void *data, size_t len) override;

  /**
   * Webrtc transport implementation to send and receive RTCP packet.
   * VideoConduit registers itself as ExternalTransport to the VideoEngine
   */
  virtual int SendRTCPPacket(int channel, const void *data, size_t len) override;


  /**
   * Webrtc External Renderer Implementation APIs.
   * Raw I420 Frames are delivred to the VideoConduit by the VideoEngine
   */
  virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int) override;

  virtual int DeliverFrame(unsigned char*, size_t, uint32_t , int64_t,
                           int64_t, void *handle) override;

  virtual int DeliverFrame(unsigned char*, size_t, uint32_t, uint32_t, uint32_t , int64_t,
                           int64_t, void *handle);

  virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;

  /**
   * Does DeliverFrame() support a null buffer and non-null handle
   * (video texture)?
   * B2G support it (when using HW video decoder with graphic buffer output).
   * XXX Investigate!  Especially for Android
   */
  virtual bool IsTextureSupported() override {
#ifdef WEBRTC_GONK
    return true;
#else
    return false;
#endif
  }

  virtual uint64_t CodecPluginID() override;

  unsigned short SendingWidth() override {
    return mSendingWidth;
  }

  unsigned short SendingHeight() override {
    return mSendingHeight;
  }

  virtual void DeleteStreams() override;
  
  unsigned int SendingMaxFs() override {
    if(mCurSendCodecConfig) {
      return mCurSendCodecConfig->mEncodingConstraints.maxFs;
    }
    return 0;
  }

  unsigned int SendingMaxFr() override {
    if(mCurSendCodecConfig) {
      return mCurSendCodecConfig->mEncodingConstraints.maxFps;
    }
    return 0;
  }

  WebrtcVideoConduit();
  virtual ~WebrtcVideoConduit();

  MediaConduitErrorCode InitMain();
  virtual MediaConduitErrorCode Init();

  int GetChannel() { return mChannel; }
  webrtc::VideoEngine* GetVideoEngine() { return mVideoEngine; }
  bool GetLocalSSRC(unsigned int* ssrc) override;
  bool SetLocalSSRC(unsigned int ssrc) override;
  bool GetRemoteSSRC(unsigned int* ssrc) override;
  bool SetLocalCNAME(const char* cname) override;
  bool GetVideoEncoderStats(double* framerateMean,
                            double* framerateStdDev,
                            double* bitrateMean,
                            double* bitrateStdDev,
                            uint32_t* droppedFrames) override;
  bool GetVideoDecoderStats(double* framerateMean,
                            double* framerateStdDev,
                            double* bitrateMean,
                            double* bitrateStdDev,
                            uint32_t* discardedPackets) override;
  bool GetAVStats(int32_t* jitterBufferDelayMs,
                  int32_t* playoutBufferDelayMs,
                  int32_t* avSyncOffsetMs) override;
  bool GetRTPStats(unsigned int* jitterMs, unsigned int* cumulativeLost) override;
  bool GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
                             uint32_t* jitterMs,
                             uint32_t* packetsReceived,
                             uint64_t* bytesReceived,
                             uint32_t* cumulativeLost,
                             int32_t* rttMs) override;
  bool GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
                           unsigned int* packetsSent,
                           uint64_t* bytesSent) override;
  uint64_t MozVideoLatencyAvg();

private:
  DISALLOW_COPY_AND_ASSIGN(WebrtcVideoConduit);

  static inline bool OnThread(nsIEventTarget *thread)
  {
    bool on;
    nsresult rv;
    rv = thread->IsOnCurrentThread(&on);

    // If the target thread has already shut down, we don't want to assert.
    if (rv != NS_ERROR_NOT_INITIALIZED) {
      MOZ_ASSERT(NS_SUCCEEDED(rv));
    }

    if (NS_WARN_IF(NS_FAILED(rv))) {
      return false;
    }
    return on;
  }

  //Local database of currently applied receive codecs
  typedef std::vector<VideoCodecConfig* > RecvCodecList;

  //Function to convert between WebRTC and Conduit codec structures
  void CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
                                webrtc::VideoCodec& cinst);

  //Checks the codec to be applied
  MediaConduitErrorCode ValidateCodecConfig(const VideoCodecConfig* codecInfo, bool send);

  //Utility function to dump recv codec database
  void DumpCodecDB() const;

  // Video Latency Test averaging filter
  void VideoLatencyUpdate(uint64_t new_sample);

  // Utility function to determine RED and ULPFEC payload types
  bool DetermineREDAndULPFECPayloadTypes(uint8_t &payload_type_red, uint8_t &payload_type_ulpfec);

  webrtc::VideoEngine* mVideoEngine;
  mozilla::ReentrantMonitor mTransportMonitor;
  RefPtr<TransportInterface> mTransmitterTransport;
  RefPtr<TransportInterface> mReceiverTransport;
  RefPtr<VideoRenderer> mRenderer;

  ScopedCustomReleasePtr<webrtc::ViEBase> mPtrViEBase;
  ScopedCustomReleasePtr<webrtc::ViECapture> mPtrViECapture;
  ScopedCustomReleasePtr<webrtc::ViECodec> mPtrViECodec;
  ScopedCustomReleasePtr<webrtc::ViENetwork> mPtrViENetwork;
  ScopedCustomReleasePtr<webrtc::ViERender> mPtrViERender;
  ScopedCustomReleasePtr<webrtc::ViERTP_RTCP> mPtrRTP;
  ScopedCustomReleasePtr<webrtc::ViEExternalCodec> mPtrExtCodec;

  webrtc::ViEExternalCapture* mPtrExtCapture;

  // Engine state we are concerned with.
  mozilla::Atomic<bool> mEngineTransmitting; //If true ==> Transmit Sub-system is up and running
  mozilla::Atomic<bool> mEngineReceiving;    // if true ==> Receive Sus-sysmtem up and running

  int mChannel; // Video Channel for this conduit
  int mCapId;   // Capturer for this conduit

  Mutex mCodecMutex; // protects mCurrSendCodecConfig
  nsAutoPtr<VideoCodecConfig> mCurSendCodecConfig;
  bool mInReconfig;

  unsigned short mLastWidth;
  unsigned short mLastHeight;
  unsigned short mSendingWidth;
  unsigned short mSendingHeight;
  unsigned short mReceivingWidth;
  unsigned short mReceivingHeight;
  unsigned int   mSendingFramerate;
  // scaled by *10 because Atomic<double/float> isn't supported
  mozilla::Atomic<int32_t, mozilla::Relaxed> mLastFramerateTenths;
  unsigned short mNumReceivingStreams;
  bool mVideoLatencyTestEnable;
  uint64_t mVideoLatencyAvg;
  uint32_t mMinBitrate;
  uint32_t mStartBitrate;
  uint32_t mMaxBitrate;
  uint32_t mMinBitrateEstimate;

  bool mRtpStreamIdEnabled;
  uint8_t mRtpStreamIdExtId;

  static const unsigned int sAlphaNum = 7;
  static const unsigned int sAlphaDen = 8;
  static const unsigned int sRoundingPadding = 1024;

  RefPtr<WebrtcAudioConduit> mSyncedTo;

  nsAutoPtr<VideoCodecConfig> mExternalSendCodec;
  nsAutoPtr<VideoCodecConfig> mExternalRecvCodec;
  nsAutoPtr<VideoEncoder> mExternalSendCodecHandle;
  nsAutoPtr<VideoDecoder> mExternalRecvCodecHandle;

  // statistics object for video codec;
  nsAutoPtr<VideoCodecStatistics> mVideoCodecStat;

  nsAutoPtr<LoadManager> mLoadManager;
  webrtc::VideoCodecMode mCodecMode;
};
} // end namespace

#endif