Moving to use data files in resources, for ACM.

The files are removed in a separate CL.

BUG=issue737

Review URL: https://webrtc-codereview.appspot.com/737004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2625 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2012-08-17 10:38:28 +00:00
parent c3115837ed
commit ba46804417
15 changed files with 222 additions and 343 deletions

View File

@ -10,6 +10,7 @@
#include "EncodeDecodeTest.h"
#include <sstream>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
@ -59,20 +60,17 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
int noOfCodecs = acm->NumberOfCodecs();
int codecNo;
if (testMode == 1) {
// Set the codec, input file, and parameters for the current test.
// Open input file
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
_pcmFile.Open(file_name, 32000, "rb");
// Set the codec for the current test.
if ((testMode == 0) || (testMode == 1)) {
// Set the codec id.
codecNo = codeId;
// Use same input file for now.
char fileName[] = "./data/audio_coding/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[] = "./data/audio_coding/testfile32kHz.pcm";
_pcmFile.Open(fileName, 32000, "rb");
} else {
// Choose codec on command line.
printf("List of supported codec.\n");
for (int n = 0; n < noOfCodecs; n++) {
acm->Codec(n, sendCodec);
@ -80,8 +78,6 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
}
printf("Choose your codec:");
ASSERT_GT(scanf("%d", &codecNo), 0);
char fileName[] = "./data/audio_coding/testfile32kHz.pcm";
_pcmFile.Open(fileName, 32000, "rb");
}
acm->Codec(codecNo, sendCodec);
@ -95,8 +91,8 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
codeId);
}
_acm = acm;
}
_acm = acm;
}
void Sender::Teardown() {
_pcmFile.Close();
@ -155,30 +151,28 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
}
}
char filename[256];
_rtpStream = rtpStream;
int playSampFreq;
std::string file_name;
std::stringstream file_stream;
file_stream << webrtc::test::OutputPath() << "encodeDecode_out" <<
static_cast<int>(codeId) << ".pcm";
file_name = file_stream.str();
_rtpStream = rtpStream;
if (testMode == 1) {
playSampFreq=recvCodec.plfreq;
//output file for current run
sprintf(filename,"%s/out%dFile.pcm", webrtc::test::OutputPath().c_str(),
codeId);
_pcmFile.Open(filename, recvCodec.plfreq, "wb+");
_pcmFile.Open(file_name, recvCodec.plfreq, "wb+");
} else if (testMode == 0) {
playSampFreq=32000;
//output file for current run
sprintf(filename, "%s/encodeDecode_out%d.pcm",
webrtc::test::OutputPath().c_str(), codeId);
_pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
_pcmFile.Open(file_name, 32000, "wb+");
} else {
printf("\nValid output frequencies:\n");
printf("8000\n16000\n32000\n-1,");
printf("which means output freq equal to received signal freq");
printf("which means output frequency equal to received signal frequency");
printf("\n\nChoose output sampling frequency: ");
ASSERT_GT(scanf("%d", &playSampFreq), 0);
sprintf(filename, "%s/outFile.pcm", webrtc::test::OutputPath().c_str());
_pcmFile.Open(filename, 32000, "wb+");
file_name = webrtc::test::OutputPath() + "encodeDecode_out.pcm";
_pcmFile.Open(file_name, playSampFreq, "wb+");
}
_realPayloadSizeBytes = 0;