blob: 9f529f56cf59e308398e4dedc1be3350cb758330 [file] [log] [blame]
/*
* Copyright (c) 2011 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(
WebRtc_Word8* 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 WebRtc_Word8* 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_