| /* |
| * Copyright (c) 2012 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_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ |
| |
| #include "audio_coding_module.h" |
| #include "audio_coding_module_typedefs.h" |
| #include "engine_configurations.h" |
| #include "module_common_types.h" |
| #include "typedefs.h" |
| #include "webrtc_neteq.h" |
| #include "webrtc_vad.h" |
| |
| namespace webrtc { |
| |
| class CriticalSectionWrapper; |
| class RWLockWrapper; |
| struct CodecInst; |
| enum AudioPlayoutMode; |
| enum ACMSpeechType; |
| |
| #define MAX_NUM_SLAVE_NETEQ 1 |
| |
| class ACMNetEQ |
| { |
| public: |
| // Constructor of the class |
| ACMNetEQ(); |
| |
| // Destructor of the class. |
| ~ACMNetEQ(); |
| |
| // |
| // GetVersion() |
| // Fills the version array with the NetEQ version and updates the |
| // remainingBufferInBytes and position variables accordingly. |
| // |
| // Output: |
| // - version : An array to be filled with the version |
| // data. |
| // |
| // Input/Output: |
| // - remainingBuffInBytes : The number of free bytes at the end of |
| // the version array. |
| // - position : Position where the free space starts. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ returned an error. |
| // |
| static WebRtc_Word32 GetVersion( |
| char* version, |
| WebRtc_UWord32& remainingBuffInBytes, |
| WebRtc_UWord32& position); |
| |
| // |
| // Init() |
| // Allocates memory for NetEQ and VAD and initializes them. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ or VAD returned an error or |
| // if out of memory. |
| // |
| WebRtc_Word32 Init(); |
| |
| // |
| // RecIn() |
| // Gives the payload to NetEQ. |
| // |
| // Input: |
| // - incomingPayload : Incoming audio payload. |
| // - payloadLength : Length of incoming audio payload. |
| // - rtpInfo : RTP header for the incoming payload containing |
| // information about payload type, sequence number, |
| // timestamp, ssrc and marker bit. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 RecIn( |
| const WebRtc_Word8* incomingPayload, |
| const WebRtc_Word32 payloadLength, |
| const WebRtcRTPHeader& rtpInfo); |
| |
| // |
| // RecOut() |
| // Asks NetEQ for 10 ms of decoded audio. |
| // |
| // Input: |
| // -audioFrame : an audio frame were output data and |
| // associated parameters are written to. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ returned an error. |
| // |
| WebRtc_Word32 RecOut( |
| AudioFrame& audioFrame); |
| |
| // |
| // AddCodec() |
| // Adds a new codec to the NetEQ codec database. |
| // |
| // Input: |
| // - codecDef : The codec to be added. |
| // - toMaster : true if the codec has to be added to Master |
| // NetEq, otherwise will be added to the Slave |
| // NetEQ. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 AddCodec( |
| WebRtcNetEQ_CodecDef *codecDef, |
| bool toMaster = true); |
| |
| // |
| // AllocatePacketBuffer() |
| // Allocates the NetEQ packet buffer. |
| // |
| // Input: |
| // - usedCodecs : An array of the codecs to be used by NetEQ. |
| // - noOfCodecs : Number of codecs in usedCodecs. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 AllocatePacketBuffer( |
| const WebRtcNetEQDecoder* usedCodecs, |
| WebRtc_Word16 noOfCodecs); |
| |
| // |
| // SetExtraDelay() |
| // Sets an delayInMS milliseconds extra delay in NetEQ. |
| // |
| // Input: |
| // - delayInMS : Extra delay in milliseconds. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 SetExtraDelay( |
| const WebRtc_Word32 delayInMS); |
| |
| // |
| // SetAVTPlayout() |
| // Enable/disable playout of AVT payloads. |
| // |
| // Input: |
| // - enable : Enable if true, disable if false. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 SetAVTPlayout( |
| const bool enable); |
| |
| // |
| // AVTPlayout() |
| // Get the current AVT playout state. |
| // |
| // Return value : True if AVT playout is enabled. |
| // False if AVT playout is disabled. |
| // |
| bool AVTPlayout() const; |
| |
| // |
| // CurrentSampFreqHz() |
| // Get the current sampling frequency in Hz. |
| // |
| // Return value : Sampling frequency in Hz. |
| // |
| WebRtc_Word32 CurrentSampFreqHz() const; |
| |
| // |
| // SetPlayoutMode() |
| // Sets the playout mode to voice or fax. |
| // |
| // Input: |
| // - mode : The playout mode to be used, voice, |
| // fax, or streaming. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 SetPlayoutMode( |
| const AudioPlayoutMode mode); |
| |
| // |
| // PlayoutMode() |
| // Get the current playout mode. |
| // |
| // Return value : The current playout mode. |
| // |
| AudioPlayoutMode PlayoutMode() const; |
| |
| // |
| // NetworkStatistics() |
| // Get the current network statistics from NetEQ. |
| // |
| // Output: |
| // - statistics : The current network statistics. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 NetworkStatistics( |
| ACMNetworkStatistics* statistics) const; |
| |
| // |
| // VADMode() |
| // Get the current VAD Mode. |
| // |
| // Return value : The current VAD mode. |
| // |
| ACMVADMode VADMode() const; |
| |
| // |
| // SetVADMode() |
| // Set the VAD mode. |
| // |
| // Input: |
| // - mode : The new VAD mode. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| WebRtc_Word16 SetVADMode( |
| const ACMVADMode mode); |
| |
| // |
| // DecodeLock() |
| // Get the decode lock used to protect decoder instances while decoding. |
| // |
| // Return value : Pointer to the decode lock. |
| // |
| RWLockWrapper* DecodeLock() const |
| { |
| return _decodeLock; |
| } |
| |
| // |
| // FlushBuffers() |
| // Flushes the NetEQ packet and speech buffers. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ returned an error. |
| // |
| WebRtc_Word32 FlushBuffers(); |
| |
| // |
| // RemoveCodec() |
| // Removes a codec from the NetEQ codec database. |
| // |
| // Input: |
| // - codecIdx : Codec to be removed. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| WebRtc_Word16 RemoveCodec( |
| WebRtcNetEQDecoder codecIdx, |
| bool isStereo = false); |
| |
| |
| // |
| // SetBackgroundNoiseMode() |
| // Set the mode of the background noise. |
| // |
| // Input: |
| // - mode : an enumerator specifying the mode of the |
| // background noise. |
| // |
| // Return value : 0 if succeeded, |
| // -1 if failed to set the mode. |
| // |
| WebRtc_Word16 SetBackgroundNoiseMode( |
| const ACMBackgroundNoiseMode mode); |
| |
| // |
| // BackgroundNoiseMode() |
| // return the mode of the background noise. |
| // |
| // Return value : The mode of background noise. |
| // |
| WebRtc_Word16 BackgroundNoiseMode( |
| ACMBackgroundNoiseMode& mode); |
| |
| void SetUniqueId( |
| WebRtc_Word32 id); |
| |
| WebRtc_Word32 PlayoutTimestamp( |
| WebRtc_UWord32& timestamp); |
| |
| void SetReceivedStereo( |
| bool receivedStereo); |
| |
| WebRtc_UWord8 NumSlaves(); |
| |
| enum JB {masterJB = 0, slaveJB = 1}; |
| |
| WebRtc_Word16 AddSlave( |
| const WebRtcNetEQDecoder* usedCodecs, |
| WebRtc_Word16 noOfCodecs); |
| |
| private: |
| // |
| // RTPPack() |
| // Creates a Word16 RTP packet out of the payload data in Word16 and |
| // a WebRtcRTPHeader. |
| // |
| // Input: |
| // - payload : Payload to be packetized. |
| // - payloadLengthW8 : Length of the payload in bytes. |
| // - rtpInfo : RTP header struct. |
| // |
| // Output: |
| // - rtpPacket : The RTP packet. |
| // |
| static void RTPPack( |
| WebRtc_Word16* rtpPacket, |
| const WebRtc_Word8* payload, |
| const WebRtc_Word32 payloadLengthW8, |
| const WebRtcRTPHeader& rtpInfo); |
| |
| void LogError( |
| const char* neteqFuncName, |
| const WebRtc_Word16 idx) const; |
| |
| WebRtc_Word16 InitByIdxSafe( |
| const WebRtc_Word16 idx); |
| |
| // EnableVAD() |
| // Enable VAD. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| WebRtc_Word16 EnableVAD(); |
| |
| WebRtc_Word16 EnableVADByIdxSafe( |
| const WebRtc_Word16 idx); |
| |
| WebRtc_Word16 AllocatePacketBufferByIdxSafe( |
| const WebRtcNetEQDecoder* usedCodecs, |
| WebRtc_Word16 noOfCodecs, |
| const WebRtc_Word16 idx); |
| |
| void* _inst[MAX_NUM_SLAVE_NETEQ + 1]; |
| void* _instMem[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| WebRtc_Word16* _netEqPacketBuffer[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| WebRtc_Word32 _id; |
| float _currentSampFreqKHz; |
| bool _avtPlayout; |
| AudioPlayoutMode _playoutMode; |
| CriticalSectionWrapper* _netEqCritSect; |
| |
| WebRtcVadInst* _ptrVADInst[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| bool _vadStatus; |
| ACMVADMode _vadMode; |
| RWLockWrapper* _decodeLock; |
| bool _isInitialized[MAX_NUM_SLAVE_NETEQ + 1]; |
| WebRtc_UWord8 _numSlaves; |
| bool _receivedStereo; |
| void* _masterSlaveInfo; |
| AudioFrame::VADActivity _previousAudioActivity; |
| WebRtc_Word32 _extraDelay; |
| |
| CriticalSectionWrapper* _callbackCritSect; |
| }; |
| |
| } //namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ |
