git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/audio_coding/main/test/Channel.h

@@ -0,0 +1,125 @@
+/*
+ *  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"
+
+
+#define MAX_NUM_PAYLOADS   50
+#define MAX_NUM_FRAMESIZES  6
+
+
+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];
+};
+
+using namespace webrtc;
+
+class Channel: public AudioPacketizationCallback
+{
+public:
+
+    Channel(
+        WebRtc_Word16 chID = -1);
+    ~Channel();
+
+    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();
+
+private:
+    void CalcStatistics(
+        WebRtcRTPHeader& rtpInfo, 
+        WebRtc_UWord16   payloadSize);
+
+    AudioCodingModule* _receiverACM;
+    WebRtc_UWord16           _seqNo;
+    // 60 msec * 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;
+};
+
+
+#endif