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

modules/audio_coding/main/test/TestAllCodecs.h

@@ -0,0 +1,94 @@
+/*
+ *  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 TEST_ALL_CODECS_H
+#define TEST_ALL_CODECS_H
+
+#include "ACMTest.h"
+#include "Channel.h"
+#include "PCMFile.h"
+
+class TestPack : public AudioPacketizationCallback
+{
+public:
+    TestPack();
+    ~TestPack();
+    
+    void RegisterReceiverACM(AudioCodingModule* acm);
+    
+    virtual 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);
+
+    WebRtc_UWord16 GetPayloadSize();
+    WebRtc_UWord32 GetTimeStampDiff();
+    void ResetPayloadSize();
+
+private:
+    AudioCodingModule* _receiverACM;
+    WebRtc_Word16            _seqNo;
+    WebRtc_UWord8            _payloadData[60 * 32 * 2 * 2]; 
+    WebRtc_UWord32           _timeStampDiff;
+    WebRtc_UWord32           _lastInTimestamp;
+    WebRtc_UWord64           _totalBytes;
+    WebRtc_UWord16           _payloadSize;
+};
+
+class TestAllCodecs : public ACMTest
+{
+public:
+    TestAllCodecs(int testMode);
+    ~TestAllCodecs();
+
+    void Perform();
+private:
+    // The default value of '-1' indicates that the registration is based only on codec name
+    // and a sampling frequncy matching is not required. This is useful for codecs which support
+    // several sampling frequency.
+    WebRtc_Word16 RegisterSendCodec(char side, 
+        char* codecName, 
+        WebRtc_Word32 sampFreqHz,
+        int rate,
+        int packSize,
+        int extraByte);
+
+    void Run(TestPack* channel);
+    void OpenOutFile(WebRtc_Word16 testNumber);
+    void DisplaySendReceiveCodec();
+
+    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);
+
+    int                     _testMode;
+
+    AudioCodingModule*      _acmA;
+    AudioCodingModule*      _acmB;
+
+    TestPack*               _channelA2B;
+
+    PCMFile                _inFileA;
+    PCMFile                _outFileB;
+    WebRtc_Word16          _testCntr;
+    WebRtc_UWord16         _packSizeSamp;
+    WebRtc_UWord16         _packSizeBytes;
+    int                    _counter;
+};
+
+
+#endif // TEST_ALL_CODECS_H
+