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

modules/audio_coding/main/test/EncodeToFileTest.cpp

@@ -0,0 +1,188 @@
+/*
+ *  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.
+ */
+
+#include "EncodeToFileTest.h"
+#include "audio_coding_module.h"
+#include "common_types.h"
+
+#ifdef WIN32
+#   include <Winsock2.h>
+#else
+#   include <arpa/inet.h>
+#endif
+
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+TestPacketization::TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency)
+:
+_frequency(frequency),
+_seqNo(0)
+{
+    _rtpStream = rtpStream;
+}
+
+TestPacketization::~TestPacketization()
+{
+}
+
+WebRtc_Word32 TestPacketization::SendData(
+    const FrameType       /* frameType */,
+    const WebRtc_UWord8   payloadType,
+    const WebRtc_UWord32  timeStamp,
+    const WebRtc_UWord8*  payloadData, 
+    const WebRtc_UWord16  payloadSize,
+    const RTPFragmentationHeader* /* fragmentation */)
+{
+    _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize, _frequency);
+    //delete [] payloadData;
+    return 1;
+}
+
+Sender::Sender()
+:
+_acm(NULL),
+//_payloadData(NULL),
+_payloadSize(0),
+_timeStamp(0)
+{
+}
+
+void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
+{
+    acm->InitializeSender();
+    struct CodecInst sendCodec;
+    int noOfCodecs = acm->NumberOfCodecs();
+    int codecNo;
+    
+    if (testMode == 1)
+    {
+        //set the codec, input file, and parameters for the current test    
+        codecNo = codeId;
+        //use same input file for now
+        char fileName[] = "./modules/audio_coding/main/test/testfile32kHz.pcm";
+        _pcmFile.Open(fileName, 32000, "rb");
+    }
+    else if (testMode == 0)
+    {
+        //set the codec, input file, and parameters for the current test    
+        codecNo = codeId;
+        acm->Codec(codecNo, sendCodec);
+        //use same input file for now
+        char fileName[] = "./modules/audio_coding/main/test/testfile32kHz.pcm";
+        _pcmFile.Open(fileName, 32000, "rb");
+    }
+    else
+    {
+        printf("List of supported codec.\n");
+        for(int n = 0; n < noOfCodecs; n++)
+        {
+            acm->Codec(n, sendCodec);
+            printf("%d %s\n", n, sendCodec.plname);
+        }
+        printf("Choose your codec:");
+    
+        scanf("%d", &codecNo);
+        char fileName[] = "./modules/audio_coding/main/test/testfile32kHz.pcm";
+        _pcmFile.Open(fileName, 32000, "rb");
+    }
+
+    acm->Codec(codecNo, sendCodec);
+    acm->RegisterSendCodec(sendCodec);
+    _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
+    if(acm->RegisterTransportCallback(_packetization) < 0)
+    {
+        printf("Registering Transport Callback failed, for run: codecId: %d: --\n",
+                codeId);
+    }
+
+    _acm = acm;
+}
+
+void Sender::Teardown()
+{
+    _pcmFile.Close();
+    delete _packetization;
+}
+
+bool Sender::Add10MsData()
+{
+    if (!_pcmFile.EndOfFile())
+    {
+        _pcmFile.Read10MsData(_audioFrame);
+        WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
+        if (ok != 0)
+        {
+            printf("Error calling Add10MsData: for run: codecId: %d\n",
+                codeId);
+            exit(1);
+        }
+        //_audioFrame._timeStamp += _pcmFile.PayloadLength10Ms();
+        return true;
+    }
+    return false;
+}
+
+bool Sender::Process()
+{
+    WebRtc_Word32 ok = _acm->Process();
+    if (ok < 0)
+    {
+        printf("Error calling Add10MsData: for run: codecId: %d\n",
+                codeId);
+        exit(1);
+    }
+    return true;
+}
+
+void Sender::Run()
+{
+    while (true)
+    {
+        if (!Add10MsData())
+        {
+            break;
+        }
+        if (!Process()) // This could be done in a processing thread
+        {
+            break;
+        }
+    }
+}
+
+EncodeToFileTest::EncodeToFileTest()
+{
+}
+
+
+void EncodeToFileTest::Perform(int fileType, int codeId, int* codePars, int testMode)
+{
+    AudioCodingModule *acm = AudioCodingModule::Create(0);
+    RTPFile rtpFile;
+    char fileName[] = "outFile.rtp";
+    rtpFile.Open(fileName, "wb+");
+    rtpFile.WriteHeader();
+
+    //for auto_test and logging
+    _sender.testMode = testMode;
+    _sender.codeId = codeId;
+
+    _sender.Setup(acm, &rtpFile);
+    struct CodecInst sendCodecInst;
+    if(acm->SendCodec(sendCodecInst) >= 0)
+    {
+        _sender.Run();
+    }
+    _sender.Teardown();
+    rtpFile.Close();
+    AudioCodingModule::Destroy(acm);
+}