blob: 375bec79d5991c32626292176ecc5793a51d66f4 [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 CHANNEL_H
#define CHANNEL_H
#include <stdio.h>
#include "audio_coding_module.h"
#include "critical_section_wrapper.h"
#include "rw_lock_wrapper.h"
namespace webrtc {
struct ACMTestFrameSizeStats
WebRtc_UWord16 frameSizeSample;
WebRtc_Word16 maxPayloadLen;
WebRtc_UWord32 numPackets;
WebRtc_UWord64 totalPayloadLenByte;
WebRtc_UWord64 totalEncodedSamples;
double rateBitPerSec;
double usageLenSec;
struct ACMTestPayloadStats
bool newPacket;
WebRtc_Word16 payloadType;
WebRtc_Word16 lastPayloadLenByte;
WebRtc_UWord32 lastTimestamp;
ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
class Channel: public AudioPacketizationCallback
WebRtc_Word16 chID = -1);
WebRtc_Word32 SendData(
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const RTPFragmentationHeader* fragmentation);
void RegisterReceiverACM(
AudioCodingModule *acm);
void ResetStats();
WebRtc_Word16 Stats(
CodecInst& codecInst,
ACMTestPayloadStats& payloadStats);
void Stats(
WebRtc_UWord32* numPackets);
void Stats(
WebRtc_UWord8* payloadLenByte,
WebRtc_UWord32* payloadType);
void PrintStats(
CodecInst& codecInst);
void SetIsStereo(bool isStereo)
_isStereo = isStereo;
WebRtc_UWord32 LastInTimestamp();
void SetFECTestWithPacketLoss(bool usePacketLoss)
_useFECTestWithPacketLoss = usePacketLoss;
double BitRate();
void CalcStatistics(
WebRtcRTPHeader& rtpInfo,
WebRtc_UWord16 payloadSize);
AudioCodingModule* _receiverACM;
WebRtc_UWord16 _seqNo;
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
CriticalSectionWrapper* _channelCritSect;
FILE* _bitStreamFile;
bool _saveBitStream;
WebRtc_Word16 _lastPayloadType;
ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
bool _isStereo;
WebRtcRTPHeader _rtpInfo;
bool _leftChannel;
WebRtc_UWord32 _lastInTimestamp;
// FEC Test variables
WebRtc_Word16 _packetLoss;
bool _useFECTestWithPacketLoss;
WebRtc_Word16 _chID;
WebRtc_UWord64 _beginTime;
WebRtc_UWord64 _totalBytes;
} // namespace webrtc