git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
446
modules/audio_coding/main/source/acm_neteq.h
Normal file
446
modules/audio_coding/main/source/acm_neteq.h
Normal file
@@ -0,0 +1,446 @@
|
||||
/*
|
||||
* 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 ACM_NETEQ_H
|
||||
#define 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(
|
||||
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;
|
||||
|
||||
//
|
||||
// JitterStatistics()
|
||||
// Get the current jitter statistics from NetEQ.
|
||||
//
|
||||
// Output:
|
||||
// - jitterStatistics : The current jitter statistics.
|
||||
//
|
||||
// Return value : 0 if ok.
|
||||
// <0 if NetEQ returned an error.
|
||||
//
|
||||
WebRtc_Word32 JitterStatistics(
|
||||
ACMJitterStatistics* jitterStatistics) const;
|
||||
|
||||
//
|
||||
// PreferredBufferSize()
|
||||
// Get the currently preferred buffer size from NetEQ.
|
||||
//
|
||||
// Output:
|
||||
// - prefBufSize : The optimal buffer size for the current network
|
||||
// conditions.
|
||||
//
|
||||
// Return value : 0 if ok.
|
||||
// <0 if NetEQ returned an error.
|
||||
//
|
||||
WebRtc_Word32 PreferredBufferSize(
|
||||
WebRtc_UWord16* prefBufSize) const;
|
||||
|
||||
//
|
||||
// ResetJitterStatistics()
|
||||
// Resets the NetEQ jitter statistics.
|
||||
//
|
||||
// Return value : 0 if ok.
|
||||
// <0 if NetEQ returned an error.
|
||||
//
|
||||
WebRtc_Word32 ResetJitterStatistics() const;
|
||||
|
||||
//
|
||||
// VADStatus()
|
||||
// Get the current VAD status.
|
||||
//
|
||||
// Return value : True if VAD is enabled.
|
||||
// False if VAD is disabled.
|
||||
//
|
||||
bool VADStatus() const;
|
||||
|
||||
//
|
||||
// SetVADStatus()
|
||||
// Enable/disable VAD.
|
||||
//
|
||||
// Input:
|
||||
// - enable : Enable if true, disable if false.
|
||||
//
|
||||
// Return value : 0 if ok.
|
||||
// -1 if an error occurred.
|
||||
//
|
||||
WebRtc_Word16 SetVADStatus(
|
||||
const bool status);
|
||||
|
||||
//
|
||||
// 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);
|
||||
|
||||
|
||||
//
|
||||
// Delay()
|
||||
// Get the length of the current audio buffer in milliseconds. That is
|
||||
// approximately the playout delay, which can be used for lip-synch.
|
||||
//
|
||||
// Output:
|
||||
// - currentDelayInMs : delay in audio buffer given in milliseconds
|
||||
//
|
||||
// return value : 0 if ok
|
||||
// -1 if an error occurred.
|
||||
//
|
||||
WebRtc_Word16 Delay(
|
||||
WebRtc_UWord16& currentDelayInMs) const;
|
||||
|
||||
//
|
||||
// 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(
|
||||
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);
|
||||
|
||||
WebRtc_Word16 EnableVADByIdxSafe(
|
||||
const WebRtc_Word16 idx);
|
||||
|
||||
WebRtc_Word16 AllocatePacketBufferByIdxSafe(
|
||||
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;
|
||||
|
||||
CriticalSectionWrapper* _callbackCritSect;
|
||||
};
|
||||
|
||||
} //namespace webrtc
|
||||
|
||||
#endif //ACM_NETEQ_H
|
||||
Reference in New Issue
Block a user