Review URL: http://webrtc-codereview.appspot.com/56003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@181 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -69,7 +69,7 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
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//set the codec, input file, and parameters for the current test
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codecNo = codeId;
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//use same input file for now
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char fileName[] = "./modules/audio_coding/main/test/testfile32kHz.pcm";
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char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
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_pcmFile.Open(fileName, 32000, "rb");
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}
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else if (testMode == 0)
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@ -78,7 +78,7 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
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codecNo = codeId;
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acm->Codec(codecNo, sendCodec);
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//use same input file for now
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char fileName[] = "./modules/audio_coding/main/test/testfile32kHz.pcm";
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char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
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_pcmFile.Open(fileName, 32000, "rb");
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}
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else
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@ -92,7 +92,7 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
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printf("Choose your codec:");
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scanf("%d", &codecNo);
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char fileName[] = "./modules/audio_coding/main/test/testfile32kHz.pcm";
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char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
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_pcmFile.Open(fileName, 32000, "rb");
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}
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