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Adding separate payload types for stereo modes

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BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2012-06-01 09:27:35 +00:00
parent c2722a0e68
commit 4517585db5
23 changed files with 670 additions and 789 deletions
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51
src/modules/audio_coding/main/test/EncodeDecodeTest.cc
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@ -149,9 +149,6 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
noOfCodecs = acm->NumberOfCodecs();
for (int i = 0; i < noOfCodecs; i++) {
acm->Codec((WebRtc_UWord8) i, recvCodec);
if (!strcmp(recvCodec.plname, "CELT")) {
recvCodec.channels = 1;
}
if (acm->RegisterReceiveCodec(recvCodec) != 0) {
printf("Unable to register codec: for run: codecId: %d\n", codeId);
exit(1);
@ -220,7 +217,6 @@ bool Receiver::IncomingPacket() {
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);
@ -296,20 +292,24 @@ void EncodeDecodeTest::Perform() {
}
int numCodecs = 1;
int codePars[3]; //freq, pacsize, rate
int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test,
//for a given codec
int codePars[3]; // Frequency, packet size, rate.
int numPars[52]; // Number of codec parameters sets (freq, pacsize, rate)
// to test, for a given codec.
codePars[0] = 0;
codePars[1] = 0;
codePars[2] = 0;
AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
struct CodecInst sendCodecTmp;
numCodecs = acmTmp->NumberOfCodecs();
AudioCodingModule::Destroy(acmTmp);
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++) {
}
if (_testMode != 2) {
for (int n = 0; n < numCodecs; n++) {
acmTmp->Codec(n, sendCodecTmp);
if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
numPars[n] = 0;
@ -317,28 +317,13 @@ void EncodeDecodeTest::Perform() {
numPars[n] = 0;
} else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
numPars[n] = 0;
} else if (sendCodecTmp.channels == 2) {
numPars[n] = 0;
} else {
numPars[n] = 1;
printf("%d %s\n", n, sendCodecTmp.plname);
}
}
AudioCodingModule::Destroy(acmTmp);
} 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;
if (_testMode == 1) {
printf("%d %s\n", n, sendCodecTmp.plname);
}
}
}
} else {
@ -348,9 +333,9 @@ void EncodeDecodeTest::Perform() {
_receiver.testMode = _testMode;
//loop over all codecs:
// Loop over all mono codecs:
for (int codeId = 0; codeId < numCodecs; codeId++) {
//only encode using real encoders, not telephone-event anc cn
// Only encode using real mono encoders, not telephone-event and cng.
for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
if (_testMode == 1) {
printf("\n");