0946a56023
BUG=314 Review URL: https://webrtc-codereview.appspot.com/1271006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
126 lines
3.1 KiB
C++
126 lines
3.1 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 CHANNEL_H
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#define CHANNEL_H
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#include <stdio.h>
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#include "audio_coding_module.h"
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#include "critical_section_wrapper.h"
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#include "rw_lock_wrapper.h"
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#include "webrtc/modules/interface/module_common_types.h"
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namespace webrtc {
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#define MAX_NUM_PAYLOADS 50
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#define MAX_NUM_FRAMESIZES 6
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struct ACMTestFrameSizeStats
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{
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uint16_t frameSizeSample;
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int16_t maxPayloadLen;
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uint32_t numPackets;
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uint64_t totalPayloadLenByte;
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uint64_t totalEncodedSamples;
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double rateBitPerSec;
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double usageLenSec;
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};
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struct ACMTestPayloadStats
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{
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bool newPacket;
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int16_t payloadType;
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int16_t lastPayloadLenByte;
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uint32_t lastTimestamp;
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ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
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};
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class Channel: public AudioPacketizationCallback
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{
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public:
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Channel(
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int16_t chID = -1);
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~Channel();
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int32_t SendData(
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const FrameType frameType,
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const uint8_t payloadType,
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const uint32_t timeStamp,
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const uint8_t* payloadData,
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const uint16_t payloadSize,
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const RTPFragmentationHeader* fragmentation);
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void RegisterReceiverACM(
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AudioCodingModule *acm);
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void ResetStats();
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int16_t Stats(
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CodecInst& codecInst,
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ACMTestPayloadStats& payloadStats);
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void Stats(
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uint32_t* numPackets);
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void Stats(
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uint8_t* payloadLenByte,
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uint32_t* payloadType);
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void PrintStats(
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CodecInst& codecInst);
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void SetIsStereo(bool isStereo)
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{
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_isStereo = isStereo;
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}
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uint32_t LastInTimestamp();
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void SetFECTestWithPacketLoss(bool usePacketLoss)
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{
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_useFECTestWithPacketLoss = usePacketLoss;
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}
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double BitRate();
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private:
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void CalcStatistics(
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WebRtcRTPHeader& rtpInfo,
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uint16_t payloadSize);
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AudioCodingModule* _receiverACM;
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uint16_t _seqNo;
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// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
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uint8_t _payloadData[60 * 32 * 2 * 2];
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CriticalSectionWrapper* _channelCritSect;
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FILE* _bitStreamFile;
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bool _saveBitStream;
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int16_t _lastPayloadType;
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ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
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bool _isStereo;
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WebRtcRTPHeader _rtpInfo;
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bool _leftChannel;
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uint32_t _lastInTimestamp;
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// FEC Test variables
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int16_t _packetLoss;
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bool _useFECTestWithPacketLoss;
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uint64_t _beginTime;
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uint64_t _totalBytes;
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};
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} // namespace webrtc
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#endif
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