Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h

@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+
+#include <stdio.h>
+
+#include "ACMTest.h"
+#include "audio_coding_module.h"
+#include "RTPFile.h"
+#include "PCMFile.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+#define MAX_INCOMING_PAYLOAD 8096
+
+// 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;
+  AudioFrame _audioFrame;
+  TestPacketization* _packetization;
+};
+
+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;
+  RTPStream* _rtpStream;
+  PCMFile _pcmFile;
+  WebRtc_Word16* _playoutBuffer;
+  WebRtc_UWord16 _playoutLengthSmpls;
+  WebRtc_UWord8 _incomingPayload[MAX_INCOMING_PAYLOAD];
+  WebRtc_UWord16 _payloadSizeBytes;
+  WebRtc_UWord16 _realPayloadSizeBytes;
+  WebRtc_Word32 _frequency;
+  bool _firstTime;
+  WebRtcRTPHeader _rtpInfo;
+  WebRtc_UWord32 _nextTime;
+};
+
+class EncodeDecodeTest: public ACMTest {
+ public:
+  EncodeDecodeTest();
+  EncodeDecodeTest(int testMode);
+  virtual void Perform();
+
+  WebRtc_UWord16 _playoutFreq;
+  WebRtc_UWord8 _testMode;
+
+ private:
+  void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
+
+ protected:
+  Sender _sender;
+  Receiver _receiver;
+};
+
+} // namespace webrtc
+
+#endif