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

modules/audio_coding/main/test/EncodeToFileTest.h

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+/*
+ *  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 ENCODETOFILETEST_H
+#define ENCODETOFILETEST_H
+
+#include "ACMTest.h"
+#include "audio_coding_module.h"
+#include "typedefs.h"
+#include "RTPFile.h"
+#include "PCMFile.h"
+#include <stdio.h>
+
+using namespace webrtc;
+
+// TestPacketization callback which writes the encoded payloads to file
+class TestPacketization : public AudioPacketizationCallback
+{
+public:
+    TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
+    ~TestPacketization();
+    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);
+
+private:
+    static void MakeRTPheader(WebRtc_UWord8* rtpHeader, 
+                              WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
+                              WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
+    RTPStream*      _rtpStream;
+    WebRtc_Word32    _frequency;
+    WebRtc_Word16     _seqNo;
+};
+
+class Sender
+{
+public:
+    Sender();
+    void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
+    void Teardown();
+    void Run();
+    bool Add10MsData();
+    bool Process();
+
+    //for auto_test and logging
+    WebRtc_UWord8             testMode;
+    WebRtc_UWord8             codeId;
+
+private:
+    AudioCodingModule*  _acm;
+    PCMFile             _pcmFile;
+    //WebRtc_Word16*    _payloadData;
+    AudioFrame          _audioFrame;
+    WebRtc_UWord16      _payloadSize;
+    WebRtc_UWord32      _timeStamp;
+    TestPacketization*  _packetization;
+};
+
+// Test class
+class EncodeToFileTest : public ACMTest
+{
+public:
+    EncodeToFileTest();
+    virtual void Perform(int fileType, int codeId, int* codePars, int testMode);
+protected:
+    Sender _sender;
+};
+
+#endif