48af652ea5
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2. BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2206004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
400 lines
11 KiB
C++
400 lines
11 KiB
C++
/*
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* Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
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#include "webrtc/common_audio/vad/include/webrtc_vad.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/typedefs.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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namespace acm1 {
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#define MAX_NUM_SLAVE_NETEQ 1
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class ACMNetEQ {
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public:
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enum JitterBuffer {
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kMasterJb = 0,
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kSlaveJb = 1
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};
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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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// 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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int32_t 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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// - incoming_payload : Incoming audio payload.
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// - length_payload : Length of incoming audio payload.
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// - rtp_info : 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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// - receive_timestamp : received timestamp.
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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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int32_t RecIn(const uint8_t* incoming_payload,
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const int32_t length_payload,
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const WebRtcRTPHeader& rtp_info,
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uint32_t receive_timestamp);
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//
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// RecIn()
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// Insert a sync payload to NetEq. Should only be called if |av_sync_| is
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// enabled;
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//
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// Input:
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// - rtp_info : 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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// - receive_timestamp : received timestamp.
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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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int RecIn(const WebRtcRTPHeader& rtp_info, uint32_t receive_timestamp);
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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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// -audio_frame : 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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int32_t RecOut(AudioFrame& audio_frame);
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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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// - codec_def : The codec to be added.
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// - to_master : 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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int32_t AddCodec(WebRtcNetEQ_CodecDef *codec_def,
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bool to_master = 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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// - used_codecs : An array of the codecs to be used by NetEQ.
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// - num_codecs : Number of codecs in used_codecs.
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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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int32_t AllocatePacketBuffer(const WebRtcNetEQDecoder* used_codecs,
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int16_t num_codecs);
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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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int32_t SetAVTPlayout(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 avt_playout() 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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int32_t 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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int32_t SetPlayoutMode(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 playout_mode() 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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int32_t NetworkStatistics(ACMNetworkStatistics* statistics) const;
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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 vad_mode() 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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int16_t SetVADMode(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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return decode_lock_;
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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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int32_t 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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// - codec_idx : 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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int16_t RemoveCodec(WebRtcNetEQDecoder codec_idx,
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bool is_stereo = false);
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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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int16_t SetBackgroundNoiseMode(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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int16_t BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
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void set_id(int32_t id);
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int32_t PlayoutTimestamp(uint32_t& timestamp);
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void set_received_stereo(bool received_stereo);
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uint8_t num_slaves();
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// Delete all slaves.
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void RemoveSlaves();
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int16_t AddSlave(const WebRtcNetEQDecoder* used_codecs,
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int16_t num_codecs);
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void BufferSpec(int& num_packets, int& size_bytes, int& overhead_bytes) {
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num_packets = min_of_max_num_packets_;
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size_bytes = min_of_buffer_size_bytes_;
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overhead_bytes = per_packet_overhead_bytes_;
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}
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//
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// Set AV-sync mode.
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//
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void EnableAVSync(bool enable);
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//
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// Get sequence number and timestamp of the last decoded RTP.
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//
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bool DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const;
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//
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// Set a minimum delay in NetEq. Unless channel condition dictates a longer
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// delay, the given delay is maintained by NetEq.
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//
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int SetMinimumDelay(int minimum_delay_ms);
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//
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// Set a maximum delay in NetEq.
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//
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int SetMaximumDelay(int maximum_delay_ms);
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//
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// The shortest latency, in milliseconds, required by jitter buffer. This
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// is computed based on inter-arrival times and playout mode of NetEq. The
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// actual delay is the maximum of least-required-delay and the minimum-delay
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// specified by SetMinumumPlayoutDelay() API.
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//
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int LeastRequiredDelayMs() const ;
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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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// - payload_length_bytes : Length of the payload in bytes.
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// - rtp_info : RTP header structure.
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//
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// Output:
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// - rtp_packet : The RTP packet.
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//
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static void RTPPack(int16_t* rtp_packet, const int8_t* payload,
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const int32_t payload_length_bytes,
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const WebRtcRTPHeader& rtp_info);
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void LogError(const char* neteq_func_name, const int16_t idx) const;
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int16_t InitByIdxSafe(const int16_t idx);
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//
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// EnableVAD()
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// Enable VAD.
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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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int16_t EnableVAD();
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int16_t EnableVADByIdxSafe(const int16_t idx);
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int16_t AllocatePacketBufferByIdxSafe(
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const WebRtcNetEQDecoder* used_codecs,
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int16_t num_codecs,
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const int16_t idx);
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// Delete the NetEQ corresponding to |index|.
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void RemoveNetEQSafe(int index);
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void RemoveSlavesSafe();
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void* inst_[MAX_NUM_SLAVE_NETEQ + 1];
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void* inst_mem_[MAX_NUM_SLAVE_NETEQ + 1];
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int16_t* neteq_packet_buffer_[MAX_NUM_SLAVE_NETEQ + 1];
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int32_t id_;
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float current_samp_freq_khz_;
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bool avt_playout_;
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AudioPlayoutMode playout_mode_;
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CriticalSectionWrapper* neteq_crit_sect_;
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WebRtcVadInst* ptr_vadinst_[MAX_NUM_SLAVE_NETEQ + 1];
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bool vad_status_;
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ACMVADMode vad_mode_;
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RWLockWrapper* decode_lock_;
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bool is_initialized_[MAX_NUM_SLAVE_NETEQ + 1];
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uint8_t num_slaves_;
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bool received_stereo_;
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void* master_slave_info_;
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AudioFrame::VADActivity previous_audio_activity_;
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CriticalSectionWrapper* callback_crit_sect_;
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// Minimum of "max number of packets," among all NetEq instances.
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int min_of_max_num_packets_;
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// Minimum of buffer-size among all NetEq instances.
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int min_of_buffer_size_bytes_;
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int per_packet_overhead_bytes_;
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// Keep track of AV-sync. Just used to set the slave when a slave is added.
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bool av_sync_;
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int minimum_delay_ms_;
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int maximum_delay_ms_;
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};
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} // namespace acm1
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
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