michael@0: /* This Source Code Form is subject to the terms of the Mozilla Public
michael@0:  * License, v. 2.0. If a copy of the MPL was not distributed with this file,
michael@0:  * You can obtain one at http://mozilla.org/MPL/2.0/. */
michael@0: 
michael@0: 
michael@0: #ifndef AUDIO_SESSION_H_
michael@0: #define AUDIO_SESSION_H_
michael@0: 
michael@0: #include "mozilla/Attributes.h"
michael@0: #include "mozilla/TimeStamp.h"
michael@0: #include "nsTArray.h"
michael@0: 
michael@0: #include "MediaConduitInterface.h"
michael@0: #include "MediaEngineWrapper.h"
michael@0: 
michael@0: // Audio Engine Includes
michael@0: #include "webrtc/common_types.h"
michael@0: #include "webrtc/voice_engine/include/voe_base.h"
michael@0: #include "webrtc/voice_engine/include/voe_volume_control.h"
michael@0: #include "webrtc/voice_engine/include/voe_codec.h"
michael@0: #include "webrtc/voice_engine/include/voe_file.h"
michael@0: #include "webrtc/voice_engine/include/voe_network.h"
michael@0: #include "webrtc/voice_engine/include/voe_external_media.h"
michael@0: #include "webrtc/voice_engine/include/voe_audio_processing.h"
michael@0: #include "webrtc/voice_engine/include/voe_video_sync.h"
michael@0: #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
michael@0: //Some WebRTC types for short notations
michael@0:  using webrtc::VoEBase;
michael@0:  using webrtc::VoENetwork;
michael@0:  using webrtc::VoECodec;
michael@0:  using webrtc::VoEExternalMedia;
michael@0:  using webrtc::VoEAudioProcessing;
michael@0:  using webrtc::VoEVideoSync;
michael@0:  using webrtc::VoERTP_RTCP;
michael@0: /** This file hosts several structures identifying different aspects
michael@0:  * of a RTP Session.
michael@0:  */
michael@0: namespace mozilla {
michael@0: // Helper function
michael@0: 
michael@0: DOMHighResTimeStamp
michael@0: NTPtoDOMHighResTimeStamp(uint32_t ntpHigh, uint32_t ntpLow);
michael@0: 
michael@0: /**
michael@0:  * Concrete class for Audio session. Hooks up
michael@0:  *  - media-source and target to external transport
michael@0:  */
michael@0: class WebrtcAudioConduit:public AudioSessionConduit
michael@0: 	      		            ,public webrtc::Transport
michael@0: {
michael@0: public:
michael@0:   //VoiceEngine defined constant for Payload Name Size.
michael@0:   static const unsigned int CODEC_PLNAME_SIZE;
michael@0: 
michael@0:   /**
michael@0:    * APIs used by the registered external transport to this Conduit to
michael@0:    * feed in received RTP Frames to the VoiceEngine for decoding
michael@0:    */
michael@0:   virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len);
michael@0: 
michael@0:   /**
michael@0:    * APIs used by the registered external transport to this Conduit to
michael@0:    * feed in received RTCP Frames to the VoiceEngine for decoding
michael@0:    */
michael@0:   virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len);
michael@0: 
michael@0:   /**
michael@0:    * Function to configure send codec for the audio session
michael@0:    * @param sendSessionConfig: CodecConfiguration
michael@0:    * @result: On Success, the audio engine is configured with passed in codec for send
michael@0:    *          On failure, audio engine transmit functionality is disabled.
michael@0:    * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
michael@0:    *        transmission sub-system on the engine.
michael@0:    */
michael@0:   virtual MediaConduitErrorCode ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig);
michael@0:   /**
michael@0:    * Function to configure list of receive codecs for the audio session
michael@0:    * @param sendSessionConfig: CodecConfiguration
michael@0:    * @result: On Success, the audio engine is configured with passed in codec for send
michael@0:    *          Also the playout is enabled.
michael@0:    *          On failure, audio engine transmit functionality is disabled.
michael@0:    * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
michael@0:    *        transmission sub-system on the engine.
michael@0:    */
michael@0:   virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
michael@0:     const std::vector<AudioCodecConfig* >& codecConfigList);
michael@0:   /**
michael@0:    * Function to enable the audio level extension
michael@0:    * @param enabled: enable extension
michael@0:    */
michael@0:   virtual MediaConduitErrorCode EnableAudioLevelExtension(bool enabled, uint8_t id);
michael@0: 
michael@0:   /**
michael@0:    * Register External Transport to this Conduit. RTP and RTCP frames from the VoiceEngine
michael@0:    * shall be passed to the registered transport for transporting externally.
michael@0:    */
michael@0:   virtual MediaConduitErrorCode AttachTransport(mozilla::RefPtr<TransportInterface> aTransport);
michael@0:   /**
michael@0:    * Function to deliver externally captured audio sample for encoding and transport
michael@0:    * @param audioData [in]: Pointer to array containing a frame of audio
michael@0:    * @param lengthSamples [in]: Length of audio frame in samples in multiple of 10 milliseconds
michael@0:    *                             Ex: Frame length is 160, 320, 440 for 16, 32, 44 kHz sampling rates
michael@0:                                     respectively.
michael@0:                                     audioData[] should be of lengthSamples in size
michael@0:                                     say, for 16kz sampling rate, audioData[] should contain 160
michael@0:                                     samples of 16-bits each for a 10m audio frame.
michael@0:    * @param samplingFreqHz [in]: Frequency/rate of the sampling in Hz ( 16000, 32000 ...)
michael@0:    * @param capture_delay [in]:  Approx Delay from recording until it is delivered to VoiceEngine
michael@0:                                  in milliseconds.
michael@0:    * NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked
michael@0:    *       This ensures the inserted audio-samples can be transmitted by the conduit
michael@0:    *
michael@0:    */
michael@0:   virtual MediaConduitErrorCode SendAudioFrame(const int16_t speechData[],
michael@0:                                                int32_t lengthSamples,
michael@0:                                                int32_t samplingFreqHz,
michael@0:                                                int32_t capture_time);
michael@0: 
michael@0:   /**
michael@0:    * Function to grab a decoded audio-sample from the media engine for rendering
michael@0:    * / playoutof length 10 milliseconds.
michael@0:    *
michael@0:    * @param speechData [in]: Pointer to a array to which a 10ms frame of audio will be copied
michael@0:    * @param samplingFreqHz [in]: Frequency of the sampling for playback in Hertz (16000, 32000,..)
michael@0:    * @param capture_delay [in]: Estimated Time between reading of the samples to rendering/playback
michael@0:    * @param lengthSamples [out]: Will contain length of the audio frame in samples at return.
michael@0:                                  Ex: A value of 160 implies 160 samples each of 16-bits was copied
michael@0:                                      into speechData
michael@0:    * NOTE: This function should be invoked every 10 milliseconds for the best
michael@0:    *          peformance
michael@0:    * NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can be invoked
michael@0:    *       This ensures the decoded samples are ready for reading and playout is enabled.
michael@0:    *
michael@0:    */
michael@0:    virtual MediaConduitErrorCode GetAudioFrame(int16_t speechData[],
michael@0:                                               int32_t samplingFreqHz,
michael@0:                                               int32_t capture_delay,
michael@0:                                               int& lengthSamples);
michael@0: 
michael@0: 
michael@0:   /**
michael@0:    * Webrtc transport implementation to send and receive RTP packet.
michael@0:    * AudioConduit registers itself as ExternalTransport to the VoiceEngine
michael@0:    */
michael@0:   virtual int SendPacket(int channel, const void *data, int len) ;
michael@0: 
michael@0:   /**
michael@0:    * Webrtc transport implementation to send and receive RTCP packet.
michael@0:    * AudioConduit registers itself as ExternalTransport to the VoiceEngine
michael@0:    */
michael@0:   virtual int SendRTCPPacket(int channel, const void *data, int len) ;
michael@0: 
michael@0: 
michael@0: 
michael@0:   WebrtcAudioConduit():
michael@0:                       mOtherDirection(nullptr),
michael@0:                       mShutDown(false),
michael@0:                       mVoiceEngine(nullptr),
michael@0:                       mTransport(nullptr),
michael@0:                       mEngineTransmitting(false),
michael@0:                       mEngineReceiving(false),
michael@0:                       mChannel(-1),
michael@0:                       mCurSendCodecConfig(nullptr),
michael@0:                       mCaptureDelay(150),
michael@0: #ifdef MOZILLA_INTERNAL_API
michael@0:                       mLastTimestamp(0),
michael@0: #endif // MOZILLA_INTERNAL_API
michael@0:                       mSamples(0),
michael@0:                       mLastSyncLog(0)
michael@0:   {
michael@0:   }
michael@0: 
michael@0:   virtual ~WebrtcAudioConduit();
michael@0: 
michael@0:   MediaConduitErrorCode Init(WebrtcAudioConduit *other);
michael@0: 
michael@0:   int GetChannel() { return mChannel; }
michael@0:   webrtc::VoiceEngine* GetVoiceEngine() { return mVoiceEngine; }
michael@0:   bool GetLocalSSRC(unsigned int* ssrc);
michael@0:   bool GetRemoteSSRC(unsigned int* ssrc);
michael@0:   bool GetAVStats(int32_t* jitterBufferDelayMs,
michael@0:                   int32_t* playoutBufferDelayMs,
michael@0:                   int32_t* avSyncOffsetMs);
michael@0:   bool GetRTPStats(unsigned int* jitterMs, unsigned int* cumulativeLost);
michael@0:   bool GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
michael@0:                              uint32_t* jitterMs,
michael@0:                              uint32_t* packetsReceived,
michael@0:                              uint64_t* bytesReceived,
michael@0:                              uint32_t *cumulativeLost,
michael@0:                              int32_t* rttMs);
michael@0:   bool GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
michael@0:                            unsigned int* packetsSent,
michael@0:                            uint64_t* bytesSent);
michael@0: 
michael@0: private:
michael@0:   WebrtcAudioConduit(const WebrtcAudioConduit& other) MOZ_DELETE;
michael@0:   void operator=(const WebrtcAudioConduit& other) MOZ_DELETE;
michael@0: 
michael@0:   //Local database of currently applied receive codecs
michael@0:   typedef std::vector<AudioCodecConfig* > RecvCodecList;
michael@0: 
michael@0:   //Function to convert between WebRTC and Conduit codec structures
michael@0:   bool CodecConfigToWebRTCCodec(const AudioCodecConfig* codecInfo,
michael@0:                                 webrtc::CodecInst& cinst);
michael@0: 
michael@0:   //Checks if given sampling frequency is supported
michael@0:   bool IsSamplingFreqSupported(int freq) const;
michael@0: 
michael@0:   //Generate block size in sample lenght for a given sampling frequency
michael@0:   unsigned int GetNum10msSamplesForFrequency(int samplingFreqHz) const;
michael@0: 
michael@0:   // Function to copy a codec structure to Conduit's database
michael@0:   bool CopyCodecToDB(const AudioCodecConfig* codecInfo);
michael@0: 
michael@0:   // Functions to verify if the codec passed is already in
michael@0:   // conduits database
michael@0:   bool CheckCodecForMatch(const AudioCodecConfig* codecInfo) const;
michael@0:   bool CheckCodecsForMatch(const AudioCodecConfig* curCodecConfig,
michael@0:                            const AudioCodecConfig* codecInfo) const;
michael@0:   //Checks the codec to be applied
michael@0:   MediaConduitErrorCode ValidateCodecConfig(const AudioCodecConfig* codecInfo, bool send) const;
michael@0: 
michael@0:   //Utility function to dump recv codec database
michael@0:   void DumpCodecDB() const;
michael@0: 
michael@0:   // The two sides of a send/receive pair of conduits each keep a pointer to the other.
michael@0:   // The also share a single VoiceEngine and mChannel.  Shutdown must be coordinated
michael@0:   // carefully to avoid double-freeing or accessing after one frees.
michael@0:   WebrtcAudioConduit*  mOtherDirection;
michael@0:   // Other side has shut down our channel and related items already
michael@0:   bool mShutDown;
michael@0: 
michael@0:   // These are shared by both directions.  They're released by the last
michael@0:   // conduit to die
michael@0:   webrtc::VoiceEngine* mVoiceEngine;
michael@0:   mozilla::RefPtr<TransportInterface> mTransport;
michael@0:   ScopedCustomReleasePtr<webrtc::VoENetwork>   mPtrVoENetwork;
michael@0:   ScopedCustomReleasePtr<webrtc::VoEBase>      mPtrVoEBase;
michael@0:   ScopedCustomReleasePtr<webrtc::VoECodec>     mPtrVoECodec;
michael@0:   ScopedCustomReleasePtr<webrtc::VoEExternalMedia> mPtrVoEXmedia;
michael@0:   ScopedCustomReleasePtr<webrtc::VoEAudioProcessing> mPtrVoEProcessing;
michael@0:   ScopedCustomReleasePtr<webrtc::VoEVideoSync> mPtrVoEVideoSync;
michael@0:   ScopedCustomReleasePtr<webrtc::VoERTP_RTCP>  mPtrVoERTP_RTCP;
michael@0:   ScopedCustomReleasePtr<webrtc::VoERTP_RTCP>  mPtrRTP;
michael@0:   //engine states of our interets
michael@0:   bool mEngineTransmitting; // If true => VoiceEngine Send-subsystem is up
michael@0:   bool mEngineReceiving;    // If true => VoiceEngine Receive-subsystem is up
michael@0:                             // and playout is enabled
michael@0:   // Keep track of each inserted RTP block and the time it was inserted
michael@0:   // so we can estimate the clock time for a specific TimeStamp coming out
michael@0:   // (for when we send data to MediaStreamTracks).  Blocks are aged out as needed.
michael@0:   struct Processing {
michael@0:     TimeStamp mTimeStamp;
michael@0:     uint32_t mRTPTimeStamp; // RTP timestamps received
michael@0:   };
michael@0:   nsAutoTArray<Processing,8> mProcessing;
michael@0: 
michael@0:   int mChannel;
michael@0:   RecvCodecList    mRecvCodecList;
michael@0:   AudioCodecConfig* mCurSendCodecConfig;
michael@0: 
michael@0:   // Current "capture" delay (really output plus input delay)
michael@0:   int32_t mCaptureDelay;
michael@0: 
michael@0: #ifdef MOZILLA_INTERNAL_API
michael@0:   uint32_t mLastTimestamp;
michael@0: #endif // MOZILLA_INTERNAL_API
michael@0: 
michael@0:   uint32_t mSamples;
michael@0:   uint32_t mLastSyncLog;
michael@0: };
michael@0: 
michael@0: } // end namespace
michael@0: 
michael@0: #endif