447 lines
13 KiB
C++
447 lines
13 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef ACM_NETEQ_H
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#define ACM_NETEQ_H
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#include "audio_coding_module.h"
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#include "audio_coding_module_typedefs.h"
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#include "engine_configurations.h"
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#include "module_common_types.h"
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#include "typedefs.h"
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#include "webrtc_neteq.h"
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#include "webrtc_vad.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RWLockWrapper;
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struct CodecInst;
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enum AudioPlayoutMode;
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enum ACMSpeechType;
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#define MAX_NUM_SLAVE_NETEQ 1
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class ACMNetEQ
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{
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public:
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// Constructor of the class
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ACMNetEQ();
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// Destructor of the class.
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~ACMNetEQ();
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//
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// GetVersion()
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// Fills the version array with the NetEQ version and updates the
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// remainingBufferInBytes and position variables accordingly.
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//
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// Output:
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// - version : An array to be filled with the version
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// data.
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//
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// Input/Output:
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// - remainingBuffInBytes : The number of free bytes at the end of
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// the version array.
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// - position : Position where the free space starts.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ returned an error.
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//
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static WebRtc_Word32 GetVersion(
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WebRtc_Word8* version,
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WebRtc_UWord32& remainingBuffInBytes,
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WebRtc_UWord32& position);
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//
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// Init()
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// Allocates memory for NetEQ and VAD and initializes them.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ or VAD returned an error or
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// if out of memory.
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//
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WebRtc_Word32 Init();
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//
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// RecIn()
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// Gives the payload to NetEQ.
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//
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// Input:
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// - incomingPayload : Incoming audio payload.
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// - payloadLength : Length of incoming audio payload.
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// - rtpInfo : RTP header for the incoming payload containing
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// information about payload type, sequence number,
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// timestamp, ssrc and marker bit.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 RecIn(
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const WebRtc_Word8* incomingPayload,
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const WebRtc_Word32 payloadLength,
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const WebRtcRTPHeader& rtpInfo);
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//
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// RecOut()
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// Asks NetEQ for 10 ms of decoded audio.
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//
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// Input:
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// -audioFrame : an audio frame were output data and
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// associated parameters are written to.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ returned an error.
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//
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WebRtc_Word32 RecOut(
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AudioFrame& audioFrame);
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//
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// AddCodec()
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// Adds a new codec to the NetEQ codec database.
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//
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// Input:
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// - codecDef : The codec to be added.
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// - toMaster : true if the codec has to be added to Master
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// NetEq, otherwise will be added to the Slave
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// NetEQ.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 AddCodec(
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WebRtcNetEQ_CodecDef *codecDef,
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bool toMaster = true);
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//
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// AllocatePacketBuffer()
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// Allocates the NetEQ packet buffer.
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//
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// Input:
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// - usedCodecs : An array of the codecs to be used by NetEQ.
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// - noOfCodecs : Number of codecs in usedCodecs.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 AllocatePacketBuffer(
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WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs);
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//
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// SetExtraDelay()
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// Sets an delayInMS milliseconds extra delay in NetEQ.
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//
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// Input:
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// - delayInMS : Extra delay in milliseconds.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 SetExtraDelay(
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const WebRtc_Word32 delayInMS);
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//
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// SetAVTPlayout()
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// Enable/disable playout of AVT payloads.
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//
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// Input:
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// - enable : Enable if true, disable if false.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 SetAVTPlayout(
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const bool enable);
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//
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// AVTPlayout()
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// Get the current AVT playout state.
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//
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// Return value : True if AVT playout is enabled.
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// False if AVT playout is disabled.
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//
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bool AVTPlayout() const;
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//
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// CurrentSampFreqHz()
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// Get the current sampling frequency in Hz.
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//
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// Return value : Sampling frequency in Hz.
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//
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WebRtc_Word32 CurrentSampFreqHz() const;
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//
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// SetPlayoutMode()
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// Sets the playout mode to voice or fax.
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//
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// Input:
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// - mode : The playout mode to be used, voice,
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// fax, or streaming.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 SetPlayoutMode(
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const AudioPlayoutMode mode);
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//
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// PlayoutMode()
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// Get the current playout mode.
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//
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// Return value : The current playout mode.
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//
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AudioPlayoutMode PlayoutMode() const;
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//
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// NetworkStatistics()
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// Get the current network statistics from NetEQ.
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//
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// Output:
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// - statistics : The current network statistics.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 NetworkStatistics(
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ACMNetworkStatistics* statistics) const;
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//
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// JitterStatistics()
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// Get the current jitter statistics from NetEQ.
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//
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// Output:
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// - jitterStatistics : The current jitter statistics.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 JitterStatistics(
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ACMJitterStatistics* jitterStatistics) const;
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//
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// PreferredBufferSize()
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// Get the currently preferred buffer size from NetEQ.
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//
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// Output:
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// - prefBufSize : The optimal buffer size for the current network
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// conditions.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 PreferredBufferSize(
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WebRtc_UWord16* prefBufSize) const;
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//
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// ResetJitterStatistics()
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// Resets the NetEQ jitter statistics.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 ResetJitterStatistics() const;
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//
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// VADStatus()
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// Get the current VAD status.
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//
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// Return value : True if VAD is enabled.
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// False if VAD is disabled.
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//
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bool VADStatus() const;
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//
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// SetVADStatus()
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// Enable/disable VAD.
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//
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// Input:
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// - enable : Enable if true, disable if false.
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//
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// Return value : 0 if ok.
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// -1 if an error occurred.
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//
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WebRtc_Word16 SetVADStatus(
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const bool status);
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//
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// VADMode()
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// Get the current VAD Mode.
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//
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// Return value : The current VAD mode.
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//
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ACMVADMode VADMode() const;
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//
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// SetVADMode()
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// Set the VAD mode.
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//
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// Input:
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// - mode : The new VAD mode.
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//
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// Return value : 0 if ok.
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// -1 if an error occurred.
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//
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WebRtc_Word16 SetVADMode(
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const ACMVADMode mode);
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//
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// DecodeLock()
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// Get the decode lock used to protect decoder instances while decoding.
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//
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// Return value : Pointer to the decode lock.
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//
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RWLockWrapper* DecodeLock() const
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{
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return _decodeLock;
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}
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//
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// FlushBuffers()
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// Flushes the NetEQ packet and speech buffers.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ returned an error.
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//
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WebRtc_Word32 FlushBuffers();
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//
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// RemoveCodec()
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// Removes a codec from the NetEQ codec database.
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//
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// Input:
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// - codecIdx : Codec to be removed.
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//
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// Return value : 0 if ok.
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// -1 if an error occurred.
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//
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WebRtc_Word16 RemoveCodec(
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WebRtcNetEQDecoder codecIdx,
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bool isStereo = false);
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//
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// Delay()
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// Get the length of the current audio buffer in milliseconds. That is
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// approximately the playout delay, which can be used for lip-synch.
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//
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// Output:
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// - currentDelayInMs : delay in audio buffer given in milliseconds
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//
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// return value : 0 if ok
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// -1 if an error occurred.
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//
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WebRtc_Word16 Delay(
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WebRtc_UWord16& currentDelayInMs) const;
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//
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// SetBackgroundNoiseMode()
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// Set the mode of the background noise.
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//
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// Input:
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// - mode : an enumerator specifying the mode of the
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// background noise.
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//
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// Return value : 0 if succeeded,
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// -1 if failed to set the mode.
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//
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WebRtc_Word16 SetBackgroundNoiseMode(
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const ACMBackgroundNoiseMode mode);
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//
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// BackgroundNoiseMode()
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// return the mode of the background noise.
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//
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// Return value : The mode of background noise.
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//
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WebRtc_Word16 BackgroundNoiseMode(
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ACMBackgroundNoiseMode& mode);
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void SetUniqueId(
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WebRtc_Word32 id);
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WebRtc_Word32 PlayoutTimestamp(
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WebRtc_UWord32& timestamp);
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void SetReceivedStereo(
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bool receivedStereo);
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WebRtc_UWord8 NumSlaves();
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enum JB {masterJB = 0, slaveJB = 1};
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WebRtc_Word16 AddSlave(
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WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs);
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private:
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//
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// RTPPack()
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// Creates a Word16 RTP packet out of the payload data in Word16 and
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// a WebRtcRTPHeader.
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//
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// Input:
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// - payload : Payload to be packetized.
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// - payloadLengthW8 : Length of the payload in bytes.
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// - rtpInfo : RTP header struct.
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//
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// Output:
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// - rtpPacket : The RTP packet.
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//
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static void RTPPack(
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WebRtc_Word16* rtpPacket,
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const WebRtc_Word8* payload,
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const WebRtc_Word32 payloadLengthW8,
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const WebRtcRTPHeader& rtpInfo);
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void LogError(
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const WebRtc_Word8* neteqFuncName,
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const WebRtc_Word16 idx) const;
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WebRtc_Word16 InitByIdxSafe(
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const WebRtc_Word16 idx);
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WebRtc_Word16 EnableVADByIdxSafe(
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const WebRtc_Word16 idx);
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WebRtc_Word16 AllocatePacketBufferByIdxSafe(
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WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs,
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const WebRtc_Word16 idx);
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void* _inst[MAX_NUM_SLAVE_NETEQ + 1];
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void* _instMem[MAX_NUM_SLAVE_NETEQ + 1];
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WebRtc_Word16* _netEqPacketBuffer[MAX_NUM_SLAVE_NETEQ + 1];
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WebRtc_Word32 _id;
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float _currentSampFreqKHz;
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bool _avtPlayout;
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AudioPlayoutMode _playoutMode;
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CriticalSectionWrapper* _netEqCritSect;
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WebRtcVadInst* _ptrVADInst[MAX_NUM_SLAVE_NETEQ + 1];
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bool _vadStatus;
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ACMVADMode _vadMode;
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RWLockWrapper* _decodeLock;
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bool _isInitialized[MAX_NUM_SLAVE_NETEQ + 1];
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WebRtc_UWord8 _numSlaves;
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bool _receivedStereo;
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void* _masterSlaveInfo;
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AudioFrame::VADActivity _previousAudioActivity;
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CriticalSectionWrapper* _callbackCritSect;
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};
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} //namespace webrtc
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#endif //ACM_NETEQ_H
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