git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp b/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp
new file mode 100644
index 0000000..08555da
--- /dev/null
+++ b/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp
@@ -0,0 +1,303 @@
+/*
+ * 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 "EncodeDecodeTest.h"
+#include "common_types.h"
+
+#include <stdlib.h>
+#include <string.h>
+#include "trace.h"
+#include "utility.h"
+
+Receiver::Receiver()
+:
+_playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
+_payloadSizeBytes(MAX_INCOMING_PAYLOAD)
+{
+}
+
+void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
+{
+ struct CodecInst recvCodec;
+ int noOfCodecs;
+ acm->InitializeReceiver();
+
+ noOfCodecs = acm->NumberOfCodecs();
+ for (int i=0; i < noOfCodecs; i++)
+ {
+ acm->Codec((WebRtc_UWord8)i, recvCodec);
+ if (acm->RegisterReceiveCodec(recvCodec) != 0)
+ {
+ printf("Unable to register codec: for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ }
+
+ char filename[128];
+ _rtpStream = rtpStream;
+ int playSampFreq;
+
+ if (testMode == 1)
+ {
+ playSampFreq=recvCodec.plfreq;
+ //output file for current run
+ sprintf(filename,"./modules/audio_coding/main/test/res_tests/out%dFile.pcm",codeId);
+ _pcmFile.Open(filename, recvCodec.plfreq, "wb+");
+ }
+ else if (testMode == 0)
+ {
+ playSampFreq=32000;
+ //output file for current run
+ sprintf(filename,"./modules/audio_coding/main/test/res_autotests/encodeDecode_out%d.pcm",codeId);
+ _pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
+ }
+ else
+ {
+ printf("\nValid output frequencies:\n");
+ printf("8000\n16000\n32000\n-1, which means output freq equal to received signal freq");
+ printf("\n\nChoose output sampling frequency: ");
+ scanf("%d", &playSampFreq);
+ char fileName[] = "./modules/audio_coding/main/test/outFile.pcm";
+ _pcmFile.Open(fileName, 32000, "wb+");
+ }
+
+ _realPayloadSizeBytes = 0;
+ _playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
+ _frequency = playSampFreq;
+ _acm = acm;
+ _firstTime = true;
+}
+
+void Receiver::Teardown()
+{
+ delete [] _playoutBuffer;
+ _pcmFile.Close();
+ if (testMode > 1) Trace::ReturnTrace();
+}
+
+bool Receiver::IncomingPacket()
+{
+ if (!_rtpStream->EndOfFile())
+ {
+ if (_firstTime)
+ {
+ _firstTime = false;
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload, _payloadSizeBytes, &_nextTime);
+ if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile())
+ {
+ _firstTime = true;
+ return true;
+ }
+ }
+
+ WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
+ if (ok != 0)
+ {
+ printf("Error when inserting packet to ACM, for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload, _payloadSizeBytes, &_nextTime);
+ if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile())
+ {
+ _firstTime = true;
+ }
+ }
+ return true;
+}
+
+bool Receiver::PlayoutData()
+{
+ AudioFrame audioFrame;
+
+ if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0)
+ {
+ printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ if (_playoutLengthSmpls == 0)
+ {
+ return false;
+ }
+ _pcmFile.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
+ return true;
+}
+
+void Receiver::Run()
+{
+ WebRtc_UWord8 counter500Ms = 50;
+
+ WebRtc_UWord32 clock = 0;
+
+ while (counter500Ms > 0)
+ {
+ if (clock == 0 || clock >= _nextTime)
+ {
+ IncomingPacket();
+ if (clock == 0)
+ {
+ clock = _nextTime;
+ }
+ }
+ if ((clock % 10) == 0)
+ {
+ if (!PlayoutData())
+ {
+ clock++;
+ continue;
+ }
+ }
+ if (_rtpStream->EndOfFile())
+ {
+ counter500Ms--;
+ }
+ clock++;
+ }
+}
+
+EncodeDecodeTest::EncodeDecodeTest()
+{
+ _testMode = 2;
+ Trace::CreateTrace();
+ Trace::SetTraceFile("acm_encdec_test.txt");
+}
+
+EncodeDecodeTest::EncodeDecodeTest(int testMode)
+{
+ //testMode == 0 for autotest
+ //testMode == 1 for testing all codecs/parameters
+ //testMode > 1 for specific user-input test (as it was used before)
+ _testMode = testMode;
+ if(_testMode != 0)
+ {
+ Trace::CreateTrace();
+ Trace::SetTraceFile("acm_encdec_test.txt");
+ }
+}
+void EncodeDecodeTest::Perform()
+{
+
+ if(_testMode == 0)
+ {
+ printf("Running Encode/Decode Test");
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- EncodeDecodeTest ----------");
+ }
+
+ int numCodecs = 1;
+ int codePars[3]; //freq, pacsize, rate
+ int playoutFreq[3]; //8, 16, 32k
+
+ int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test, for a given codec
+
+ codePars[0]=0;
+ codePars[1]=0;
+ codePars[2]=0;
+
+ if (_testMode == 1)
+ {
+ AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
+ struct CodecInst sendCodecTmp;
+ numCodecs = acmTmp->NumberOfCodecs();
+ printf("List of supported codec.\n");
+ for(int n = 0; n < numCodecs; n++)
+ {
+ acmTmp->Codec(n, sendCodecTmp);
+ if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
+ numPars[n] = 0;
+ } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
+ numPars[n] = 0;
+ } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
+ numPars[n] = 0;
+ } else {
+ numPars[n] = 1;
+ printf("%d %s\n", n, sendCodecTmp.plname);
+ }
+ }
+ AudioCodingModule::Destroy(acmTmp);
+ playoutFreq[1]=16000;
+ }
+ else if (_testMode == 0)
+ {
+ AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
+ numCodecs = acmTmp->NumberOfCodecs();
+ AudioCodingModule::Destroy(acmTmp);
+ struct CodecInst dummyCodec;
+
+ //chose range of testing for codecs/parameters
+ for(int i = 0 ; i < numCodecs ; i++)
+ {
+ numPars[i] = 1;
+ acmTmp->Codec(i, dummyCodec);
+ if (STR_CASE_CMP(dummyCodec.plname, "telephone-event") == 0)
+ {
+ numPars[i] = 0;
+ } else if (STR_CASE_CMP(dummyCodec.plname, "cn") == 0) {
+ numPars[i] = 0;
+ } else if (STR_CASE_CMP(dummyCodec.plname, "red") == 0) {
+ numPars[i] = 0;
+ }
+ }
+ playoutFreq[1] = 16000;
+ }
+ else
+ {
+ numCodecs = 1;
+ numPars[0] = 1;
+ playoutFreq[1]=16000;
+ }
+
+ _receiver.testMode = _testMode;
+
+ //loop over all codecs:
+ for(int codeId=0;codeId<numCodecs;codeId++)
+ {
+ //only encode using real encoders, not telephone-event anc cn
+ for(int loopPars=1;loopPars<=numPars[codeId];loopPars++)
+ {
+ if (_testMode == 1)
+ {
+ printf("\n");
+ printf("***FOR RUN: codeId: %d\n",codeId);
+ printf("\n");
+ }
+ else if (_testMode == 0)
+ {
+ printf(".");
+ }
+
+ EncodeToFileTest::Perform(1, codeId, codePars, _testMode);
+
+ AudioCodingModule *acm = AudioCodingModule::Create(10);
+ RTPFile rtpFile;
+ char fileName[] = "outFile.rtp";
+ rtpFile.Open(fileName, "rb");
+
+ _receiver.codeId = codeId;
+
+ rtpFile.ReadHeader();
+ _receiver.Setup(acm, &rtpFile);
+ _receiver.Run();
+ _receiver.Teardown();
+ rtpFile.Close();
+ AudioCodingModule::Destroy(acm);
+
+ if (_testMode == 1)
+ {
+ printf("***COMPLETED RUN FOR: codecID: %d ***\n",
+ codeId);
+ }
+ }
+ }
+ if (_testMode == 0)
+ {
+ printf("Done!\n");
+ }
+ if (_testMode == 1) Trace::ReturnTrace();
+}
+
