Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/test/EncodeDecodeTest.h

@@ -0,0 +1,123 @@
+/*
+ *  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_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+
+#include <stdio.h>
+#include <string.h>
+
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/RTPFile.h"
+#include "webrtc/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, uint16_t frequency);
+  ~TestPacketization();
+  int32_t SendData(const FrameType frameType,
+                   const uint8_t payloadType,
+                   const uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   const size_t payloadSize,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+ private:
+  static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
+                            int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
+  RTPStream* _rtpStream;
+  int32_t _frequency;
+  int16_t _seqNo;
+};
+
+class Sender {
+ public:
+  Sender();
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string in_file_name, int sample_rate, size_t channels);
+  void Teardown();
+  void Run();
+  bool Add10MsData();
+
+  //for auto_test and logging
+  uint8_t testMode;
+  uint8_t codeId;
+
+ protected:
+  AudioCodingModule* _acm;
+
+ private:
+  PCMFile _pcmFile;
+  AudioFrame _audioFrame;
+  TestPacketization* _packetization;
+};
+
+class Receiver {
+ public:
+  Receiver();
+  virtual ~Receiver() {};
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string out_file_name, size_t channels);
+  void Teardown();
+  void Run();
+  virtual bool IncomingPacket();
+  bool PlayoutData();
+
+  //for auto_test and logging
+  uint8_t codeId;
+  uint8_t testMode;
+
+ private:
+  PCMFile _pcmFile;
+  int16_t* _playoutBuffer;
+  uint16_t _playoutLengthSmpls;
+  int32_t _frequency;
+  bool _firstTime;
+
+ protected:
+  AudioCodingModule* _acm;
+  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
+  RTPStream* _rtpStream;
+  WebRtcRTPHeader _rtpInfo;
+  size_t _realPayloadSizeBytes;
+  size_t _payloadSizeBytes;
+  uint32_t _nextTime;
+};
+
+class EncodeDecodeTest : public ACMTest {
+ public:
+  EncodeDecodeTest();
+  explicit EncodeDecodeTest(int testMode);
+  void Perform() override;
+
+  uint16_t _playoutFreq;
+  uint8_t _testMode;
+
+ private:
+  std::string EncodeToFile(int fileType,
+                           int codeId,
+                           int* codePars,
+                           int testMode);
+
+ protected:
+  Sender _sender;
+  Receiver _receiver;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_