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

modules/audio_coding/main/test/EncodeDecodeTest.h

@@ -0,0 +1,64 @@
+/*
+ *  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 ENCODEDECODETEST_H
+#define ENCODEDECODETEST_H
+
+#include "EncodeToFileTest.h"
+
+#define MAX_INCOMING_PAYLOAD 8096
+#include "audio_coding_module.h"
+
+class Receiver
+{
+public:
+    Receiver();
+    void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
+    void Teardown();
+    void Run();
+    bool IncomingPacket();
+    bool PlayoutData();    
+
+    //for auto_test and logging
+    WebRtc_UWord8             codeId;
+    WebRtc_UWord8             testMode;
+
+private:
+    AudioCodingModule*    _acm;
+    bool                  _rtpEOF;
+    RTPStream*            _rtpStream;
+    PCMFile               _pcmFile;
+    WebRtc_Word16*        _playoutBuffer;
+    WebRtc_UWord16        _playoutLengthSmpls;
+    WebRtc_Word8          _incomingPayload[MAX_INCOMING_PAYLOAD];
+    WebRtc_UWord16        _payloadSizeBytes;
+    WebRtc_UWord16        _realPayloadSizeBytes;
+    WebRtc_Word32         _frequency;
+    bool                  _firstTime;
+    WebRtcRTPHeader       _rtpInfo;
+    WebRtc_UWord32        _nextTime;
+};
+
+class EncodeDecodeTest : public EncodeToFileTest
+{
+public:
+    EncodeDecodeTest();
+    EncodeDecodeTest(int testMode);
+    virtual void Perform();
+    WebRtc_UWord16            _playoutFreq;    
+    WebRtc_UWord8             _testMode;
+protected:
+    Receiver    _receiver;    
+};      
+
+
+
+#endif
+