/* 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 AUDIO_SESSION_H_ #define AUDIO_SESSION_H_ #include "mozilla/Attributes.h" #include "mozilla/TimeStamp.h" #include "nsTArray.h" #include "MediaConduitInterface.h" #include "MediaEngineWrapper.h" // Audio Engine Includes #include "webrtc/common_types.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_volume_control.h" #include "webrtc/voice_engine/include/voe_codec.h" #include "webrtc/voice_engine/include/voe_file.h" #include "webrtc/voice_engine/include/voe_network.h" #include "webrtc/voice_engine/include/voe_external_media.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_video_sync.h" #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" //Some WebRTC types for short notations using webrtc::VoEBase; using webrtc::VoENetwork; using webrtc::VoECodec; using webrtc::VoEExternalMedia; using webrtc::VoEAudioProcessing; using webrtc::VoEVideoSync; using webrtc::VoERTP_RTCP; /** This file hosts several structures identifying different aspects * of a RTP Session. */ namespace mozilla { // Helper function DOMHighResTimeStamp NTPtoDOMHighResTimeStamp(uint32_t ntpHigh, uint32_t ntpLow); /** * Concrete class for Audio session. Hooks up * - media-source and target to external transport */ class WebrtcAudioConduit:public AudioSessionConduit ,public webrtc::Transport { public: //VoiceEngine defined constant for Payload Name Size. static const unsigned int CODEC_PLNAME_SIZE; /** * APIs used by the registered external transport to this Conduit to * feed in received RTP Frames to the VoiceEngine 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 RTCP Frames to the VoiceEngine 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 send codec for the audio session * @param sendSessionConfig: CodecConfiguration * @result: On Success, the audio engine is configured with passed in codec for send * On failure, audio 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 AudioCodecConfig* codecConfig) override; /** * Function to configure list of receive codecs for the audio session * @param sendSessionConfig: CodecConfiguration * @result: On Success, the audio engine is configured with passed in codec for send * Also the playout is enabled. * On failure, audio 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& codecConfigList) override; /** * Function to enable the audio level extension * @param enabled: enable extension */ virtual MediaConduitErrorCode EnableAudioLevelExtension(bool enabled, uint8_t id) override; /** * Register External Transport to this Conduit. RTP and RTCP frames from the VoiceEngine * shall be passed to the registered transport for transporting externally. */ virtual MediaConduitErrorCode SetTransmitterTransport(RefPtr aTransport) override; virtual MediaConduitErrorCode SetReceiverTransport(RefPtr aTransport) override; /** * Function to deliver externally captured audio sample for encoding and transport * @param audioData [in]: Pointer to array containing a frame of audio * @param lengthSamples [in]: Length of audio frame in samples in multiple of 10 milliseconds * Ex: Frame length is 160, 320, 440 for 16, 32, 44 kHz sampling rates respectively. audioData[] should be of lengthSamples in size say, for 16kz sampling rate, audioData[] should contain 160 samples of 16-bits each for a 10m audio frame. * @param samplingFreqHz [in]: Frequency/rate of the sampling in Hz ( 16000, 32000 ...) * @param capture_delay [in]: Approx Delay from recording until it is delivered to VoiceEngine in milliseconds. * NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked * This ensures the inserted audio-samples can be transmitted by the conduit * */ virtual MediaConduitErrorCode SendAudioFrame(const int16_t speechData[], int32_t lengthSamples, int32_t samplingFreqHz, int32_t capture_time) override; /** * Function to grab a decoded audio-sample from the media engine for rendering * / playoutof length 10 milliseconds. * * @param speechData [in]: Pointer to a array to which a 10ms frame of audio will be copied * @param samplingFreqHz [in]: Frequency of the sampling for playback in Hertz (16000, 32000,..) * @param capture_delay [in]: Estimated Time between reading of the samples to rendering/playback * @param lengthSamples [out]: Will contain length of the audio frame in samples at return. Ex: A value of 160 implies 160 samples each of 16-bits was copied into speechData * NOTE: This function should be invoked every 10 milliseconds for the best * peformance * NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can be invoked * This ensures the decoded samples are ready for reading and playout is enabled. * */ virtual MediaConduitErrorCode GetAudioFrame(int16_t speechData[], int32_t samplingFreqHz, int32_t capture_delay, int& lengthSamples) override; /** * Webrtc transport implementation to send and receive RTP packet. * AudioConduit registers itself as ExternalTransport to the VoiceEngine */ virtual int SendPacket(int channel, const void *data, size_t len) override; /** * Webrtc transport implementation to send and receive RTCP packet. * AudioConduit registers itself as ExternalTransport to the VoiceEngine */ virtual int SendRTCPPacket(int channel, const void *data, size_t len) override; virtual uint64_t CodecPluginID() override { return 0; } WebrtcAudioConduit(): mVoiceEngine(nullptr), mTransportMonitor("WebrtcAudioConduit"), mTransmitterTransport(nullptr), mReceiverTransport(nullptr), mEngineTransmitting(false), mEngineReceiving(false), mChannel(-1), mDtmfEnabled(false), mCodecMutex("AudioConduit codec db"), mCaptureDelay(150), #if !defined(MOZILLA_EXTERNAL_LINKAGE) mLastTimestamp(0), #endif // MOZILLA_INTERNAL_API mSamples(0), mLastSyncLog(0) { } virtual ~WebrtcAudioConduit(); MediaConduitErrorCode Init(); int GetChannel() { return mChannel; } webrtc::VoiceEngine* GetVoiceEngine() { return mVoiceEngine; } bool SetLocalSSRC(unsigned int ssrc) override; bool GetLocalSSRC(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 { return false; } bool GetVideoDecoderStats(double* framerateMean, double* framerateStdDev, double* bitrateMean, double* bitrateStdDev, uint32_t* discardedPackets) override { return false; } 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; bool SetDtmfPayloadType(unsigned char type) override; bool InsertDTMFTone(int channel, int eventCode, bool outOfBand, int lengthMs, int attenuationDb) override; private: WebrtcAudioConduit(const WebrtcAudioConduit& other) = delete; void operator=(const WebrtcAudioConduit& other) = delete; //Local database of currently applied receive codecs typedef std::vector RecvCodecList; //Function to convert between WebRTC and Conduit codec structures bool CodecConfigToWebRTCCodec(const AudioCodecConfig* codecInfo, webrtc::CodecInst& cinst); //Checks if given sampling frequency is supported bool IsSamplingFreqSupported(int freq) const; //Generate block size in sample lenght for a given sampling frequency unsigned int GetNum10msSamplesForFrequency(int samplingFreqHz) const; // Function to copy a codec structure to Conduit's database bool CopyCodecToDB(const AudioCodecConfig* codecInfo); // Functions to verify if the codec passed is already in // conduits database bool CheckCodecForMatch(const AudioCodecConfig* codecInfo) const; bool CheckCodecsForMatch(const AudioCodecConfig* curCodecConfig, const AudioCodecConfig* codecInfo) const; //Checks the codec to be applied MediaConduitErrorCode ValidateCodecConfig(const AudioCodecConfig* codecInfo, bool send); //Utility function to dump recv codec database void DumpCodecDB() const; webrtc::VoiceEngine* mVoiceEngine; mozilla::ReentrantMonitor mTransportMonitor; RefPtr mTransmitterTransport; RefPtr mReceiverTransport; ScopedCustomReleasePtr mPtrVoENetwork; ScopedCustomReleasePtr mPtrVoEBase; ScopedCustomReleasePtr mPtrVoECodec; ScopedCustomReleasePtr mPtrVoEXmedia; ScopedCustomReleasePtr mPtrVoEProcessing; ScopedCustomReleasePtr mPtrVoEVideoSync; ScopedCustomReleasePtr mPtrVoERTP_RTCP; ScopedCustomReleasePtr mPtrRTP; //engine states of our interets mozilla::Atomic mEngineTransmitting; // If true => VoiceEngine Send-subsystem is up mozilla::Atomic mEngineReceiving; // If true => VoiceEngine Receive-subsystem is up // and playout is enabled // Keep track of each inserted RTP block and the time it was inserted // so we can estimate the clock time for a specific TimeStamp coming out // (for when we send data to MediaStreamTracks). Blocks are aged out as needed. struct Processing { TimeStamp mTimeStamp; uint32_t mRTPTimeStamp; // RTP timestamps received }; AutoTArray mProcessing; int mChannel; bool mDtmfEnabled; RecvCodecList mRecvCodecList; Mutex mCodecMutex; // protects mCurSendCodecConfig nsAutoPtr mCurSendCodecConfig; // Current "capture" delay (really output plus input delay) int32_t mCaptureDelay; #if !defined(MOZILLA_EXTERNAL_LINKAGE) uint32_t mLastTimestamp; #endif // MOZILLA_INTERNAL_API uint32_t mSamples; uint32_t mLastSyncLog; }; } // end namespace #endif