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Changing non-const reference arguments to pointers, ACM

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Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Committed: https://code.google.com/p/webrtc/source/detail?r=3543

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2013-02-21 10:27:48 +00:00
parent f61e02c81f
commit 7a7a008031
20 changed files with 214 additions and 207 deletions
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42
webrtc/modules/audio_coding/main/test/APITest.cc
View File

@ -182,7 +182,7 @@ APITest::SetUp()
WebRtc_Word16 numCodecs = _acmA->NumberOfCodecs();
for(WebRtc_UWord8 n = 0; n < numCodecs; n++)
{
AudioCodingModule::Codec(n, dummyCodec);
AudioCodingModule::Codec(n, &dummyCodec);
if((STR_CASE_CMP(dummyCodec.plname, "CN") == 0) &&
(dummyCodec.plfreq == 32000))
{
@ -205,7 +205,7 @@ APITest::SetUp()
// test if re-registration works;
CodecInst nextCodec;
int currentPayloadType = dummyCodec.pltype;
AudioCodingModule::Codec(n + 1, nextCodec);
AudioCodingModule::Codec(n + 1, &nextCodec);
dummyCodec.pltype = nextCodec.pltype;
if(!FixedPayloadTypeCodec(nextCodec.plname))
{
@ -218,7 +218,7 @@ APITest::SetUp()
{
// test if un-registration works;
CodecInst nextCodec;
AudioCodingModule::Codec(n + 1, nextCodec);
AudioCodingModule::Codec(n + 1, &nextCodec);
nextCodec.pltype = dummyCodec.pltype;
if(!FixedPayloadTypeCodec(nextCodec.plname))
{
@ -248,11 +248,11 @@ APITest::SetUp()
_thereIsDecoderB = true;
// Register Send Codec
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, dummyCodec);
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, &dummyCodec);
CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
_thereIsEncoderA = true;
//
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, dummyCodec);
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, &dummyCodec);
CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
_thereIsEncoderB = true;
@ -410,7 +410,7 @@ APITest::PullAudioRunA()
{
_pullEventA->Wait(100);
AudioFrame audioFrame;
if(_acmA->PlayoutData10Ms(_outFreqHzA, audioFrame) < 0)
if(_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame) < 0)
{
bool thereIsDecoder;
{
@ -438,7 +438,7 @@ APITest::PullAudioRunB()
{
_pullEventB->Wait(100);
AudioFrame audioFrame;
if(_acmB->PlayoutData10Ms(_outFreqHzB, audioFrame) < 0)
if(_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame) < 0)
{
bool thereIsDecoder;
{
@ -794,7 +794,7 @@ APITest::CheckVADStatus(char side)
if(side == 'A')
{
_acmA->VAD(dtxEnabled, vadEnabled, vadMode);
_acmA->VAD(&dtxEnabled, &vadEnabled, &vadMode);
_acmA->RegisterVADCallback(NULL);
_vadCallbackA->Reset();
_acmA->RegisterVADCallback(_vadCallbackA);
@ -838,7 +838,7 @@ APITest::CheckVADStatus(char side)
}
else
{
_acmB->VAD(dtxEnabled, vadEnabled, vadMode);
_acmB->VAD(&dtxEnabled, &vadEnabled, &vadMode);
_acmB->RegisterVADCallback(NULL);
_vadCallbackB->Reset();
@ -920,7 +920,7 @@ APITest::TestDelay(char side)
inTimestamp = myChannel->LastInTimestamp();
CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
if(!_randomTest)
{
@ -932,7 +932,7 @@ APITest::TestDelay(char side)
myEvent->Wait(1000);
inTimestamp = myChannel->LastInTimestamp();
CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
//std::cout << outTimestamp << std::endl << std::flush;
estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) /
@ -968,7 +968,7 @@ APITest::TestDelay(char side)
*myMinDelay = (rand() % 1000) + 1;
ACMNetworkStatistics networkStat;
CHECK_ERROR_MT(myACM->NetworkStatistics(networkStat));
CHECK_ERROR_MT(myACM->NetworkStatistics(&networkStat));
if(!_randomTest)
{
@ -1039,9 +1039,9 @@ APITest::TestRegisteration(char sendSide)
}
CodecInst myCodec;
if(sendACM->SendCodec(myCodec) < 0)
if(sendACM->SendCodec(&myCodec) < 0)
{
AudioCodingModule::Codec(_codecCntrA, myCodec);
AudioCodingModule::Codec(_codecCntrA, &myCodec);
}
if(!_randomTest)
@ -1332,7 +1332,7 @@ APITest::TestSendVAD(char side)
if(side == 'A')
{
AudioCodingModule::Codec(_codecCntrA, myCodec);
AudioCodingModule::Codec(_codecCntrA, &myCodec);
vad = &_sendVADA;
dtx = &_sendDTXA;
mode = &_sendVADModeA;
@ -1341,7 +1341,7 @@ APITest::TestSendVAD(char side)
}
else
{
AudioCodingModule::Codec(_codecCntrB, myCodec);
AudioCodingModule::Codec(_codecCntrB, &myCodec);
vad = &_sendVADB;
dtx = &_sendDTXB;
mode = &_sendVADModeB;
@ -1408,11 +1408,11 @@ APITest::CurrentCodec(char side)
CodecInst myCodec;
if(side == 'A')
{
_acmA->SendCodec(myCodec);
_acmA->SendCodec(&myCodec);
}
else
{
_acmB->SendCodec(myCodec);
_acmB->SendCodec(&myCodec);
}
if(!_randomTest)
@ -1493,11 +1493,11 @@ APITest::ChangeCodec(char side)
Wait(1000);
// After Initialization CN is lost, re-register them
if(AudioCodingModule::Codec("CN", myCodec, 8000, 1) >= 0)
if(AudioCodingModule::Codec("CN", &myCodec, 8000, 1) >= 0)
{
CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
}
if(AudioCodingModule::Codec("CN", myCodec, 16000, 1) >= 0)
if(AudioCodingModule::Codec("CN", &myCodec, 16000, 1) >= 0)
{
CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
}
@ -1507,7 +1507,7 @@ APITest::ChangeCodec(char side)
_writeToFile = false;
}
AudioCodingModule::Codec(*codecCntr, myCodec);
AudioCodingModule::Codec(*codecCntr, &myCodec);
} while(!STR_CASE_CMP(myCodec.plname, "CN") ||
!STR_CASE_CMP(myCodec.plname, "telephone-event") ||
!STR_CASE_CMP(myCodec.plname, "RED"));

12
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@ -73,14 +73,14 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
// Choose codec on command line.
printf("List of supported codec.\n");
for (int n = 0; n < noOfCodecs; n++) {
acm->Codec(n, sendCodec);
acm->Codec(n, &sendCodec);
printf("%d %s\n", n, sendCodec.plname);
}
printf("Choose your codec:");
ASSERT_GT(scanf("%d", &codecNo), 0);
}
acm->Codec(codecNo, sendCodec);
acm->Codec(codecNo, &sendCodec);
if (!strcmp(sendCodec.plname, "CELT")) {
sendCodec.channels = 1;
}
@ -144,7 +144,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
noOfCodecs = acm->NumberOfCodecs();
for (int i = 0; i < noOfCodecs; i++) {
acm->Codec((WebRtc_UWord8) i, recvCodec);
acm->Codec((WebRtc_UWord8) i, &recvCodec);
if (acm->RegisterReceiveCodec(recvCodec) != 0) {
printf("Unable to register codec: for run: codecId: %d\n", codeId);
exit(1);
@ -224,7 +224,7 @@ bool Receiver::IncomingPacket() {
bool Receiver::PlayoutData() {
AudioFrame audioFrame;
if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0) {
if (_acm->PlayoutData10Ms(_frequency, &audioFrame) != 0) {
printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n",
codeId);
exit(1);
@ -305,7 +305,7 @@ void EncodeDecodeTest::Perform() {
}
if (_testMode != 2) {
for (int n = 0; n < numCodecs; n++) {
acm->Codec(n, sendCodecTmp);
acm->Codec(n, &sendCodecTmp);
if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
numPars[n] = 0;
} else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
@ -381,7 +381,7 @@ void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
_sender.Setup(acm, &rtpFile);
struct CodecInst sendCodecInst;
if (acm->SendCodec(sendCodecInst) >= 0) {
if (acm->SendCodec(&sendCodecInst) >= 0) {
_sender.Run();
}
_sender.Teardown();

18
webrtc/modules/audio_coding/main/test/SpatialAudio.cc
View File

@ -82,7 +82,7 @@ SpatialAudio::Setup()
WebRtc_UWord8 num_encoders = _acmReceiver->NumberOfCodecs();
// Register all available codes as receiving codecs once more.
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
status = _acmReceiver->Codec(n, codecInst);
status = _acmReceiver->Codec(n, &codecInst);
if (status < 0) {
printf("Error in Codec(), no matching codec found");
}
@ -109,7 +109,7 @@ SpatialAudio::Perform()
Setup();
CodecInst codecInst;
_acmLeft->Codec((WebRtc_UWord8)1, codecInst);
_acmLeft->Codec((WebRtc_UWord8)1, &codecInst);
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
EncodeDecode();
@ -122,7 +122,7 @@ SpatialAudio::Perform()
while((pannCntr + 1) < NUM_PANN_COEFFS)
{
_acmLeft->Codec((WebRtc_UWord8)0, codecInst);
_acmLeft->Codec((WebRtc_UWord8)0, &codecInst);
codecInst.pacsize = 480;
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@ -131,7 +131,7 @@ SpatialAudio::Perform()
pannCntr++;
// Change codec
_acmLeft->Codec((WebRtc_UWord8)3, codecInst);
_acmLeft->Codec((WebRtc_UWord8)3, &codecInst);
codecInst.pacsize = 320;
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@ -144,11 +144,11 @@ SpatialAudio::Perform()
}
}
_acmLeft->Codec((WebRtc_UWord8)4, codecInst);
_acmLeft->Codec((WebRtc_UWord8)4, &codecInst);
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
EncodeDecode();
_acmLeft->Codec((WebRtc_UWord8)0, codecInst);
_acmLeft->Codec((WebRtc_UWord8)0, &codecInst);
codecInst.pacsize = 480;
CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
@ -200,7 +200,8 @@ SpatialAudio::EncodeDecode(
CHECK_ERROR(_acmLeft->Process());
CHECK_ERROR(_acmRight->Process());
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq,
&audioFrame));
_outFile.Write10MsData(audioFrame);
}
_inFile.Rewind();
@ -221,7 +222,8 @@ SpatialAudio::EncodeDecode()
CHECK_ERROR(_acmLeft->Process());
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq,
&audioFrame));
_outFile.Write10MsData(audioFrame);
}
_inFile.Rewind();

10
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
View File

@ -145,7 +145,7 @@ void TestAllCodecs::Perform() {
uint8_t num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (uint8_t n = 0; n < num_encoders; n++) {
acm_b_->Codec(n, my_codec_param);
acm_b_->Codec(n, &my_codec_param);
if (!strcmp(my_codec_param.plname, "opus")) {
my_codec_param.channels = 1;
}
@ -752,7 +752,7 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
// Get all codec parameters before registering
CodecInst my_codec_param;
CHECK_ERROR(AudioCodingModule::Codec(codec_name, my_codec_param,
CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
sampling_freq_hz, 1));
my_codec_param.rate = rate;
my_codec_param.pacsize = packet_size;
@ -795,7 +795,7 @@ void TestAllCodecs::Run(TestPack* channel) {
}
// Run received side of ACM.
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, audio_frame));
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame));
// Write output speech to file.
outfile_b_.Write10MsData(audio_frame.data_,
@ -824,9 +824,9 @@ void TestAllCodecs::OpenOutFile(int test_number) {
void TestAllCodecs::DisplaySendReceiveCodec() {
CodecInst my_codec_param;
acm_a_->SendCodec(my_codec_param);
acm_a_->SendCodec(&my_codec_param);
printf("%s -> ", my_codec_param.plname);
acm_b_->ReceiveCodec(my_codec_param);
acm_b_->ReceiveCodec(&my_codec_param);
printf("%s\n", my_codec_param.plname);
}

10
webrtc/modules/audio_coding/main/test/TestFEC.cc
View File

@ -79,7 +79,7 @@ void TestFEC::Perform()
}
for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
{
_acmB->Codec(n, myCodecParam);
_acmB->Codec(n, &myCodecParam);
if(_testMode != 0)
{
printf("%s\n", myCodecParam.plname);
@ -553,7 +553,7 @@ WebRtc_Word16 TestFEC::RegisterSendCodec(char side, char* codecName, WebRtc_Word
}
CodecInst myCodecParam;
CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam,
CHECK_ERROR(AudioCodingModule::Codec(codecName, &myCodecParam,
samplingFreqHz, 1));
CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
@ -575,7 +575,7 @@ void TestFEC::Run()
_inFileA.Read10MsData(audioFrame);
CHECK_ERROR(_acmA->Add10MsData(audioFrame));
CHECK_ERROR(_acmA->Process());
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
msecPassed += 10;
if(msecPassed >= 1000)
@ -616,9 +616,9 @@ void TestFEC::OpenOutFile(WebRtc_Word16 test_number) {
void TestFEC::DisplaySendReceiveCodec()
{
CodecInst myCodecParam;
_acmA->SendCodec(myCodecParam);
_acmA->SendCodec(&myCodecParam);
printf("%s -> ", myCodecParam.plname);
_acmB->ReceiveCodec(myCodecParam);
_acmB->ReceiveCodec(&myCodecParam);
printf("%s\n", myCodecParam.plname);
}

30
webrtc/modules/audio_coding/main/test/TestStereo.cc
View File

@ -182,19 +182,19 @@ void TestStereo::Perform() {
WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, my_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
}
// Test that unregister all receive codecs works.
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, my_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
EXPECT_EQ(0, acm_b_->UnregisterReceiveCodec(my_codec_param.pltype));
}
// Register all available codes as receiving codecs once more.
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, my_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
}
@ -222,12 +222,12 @@ void TestStereo::Perform() {
// Continue with setting a stereo codec as send codec and verify that
// VAD/DTX gets turned off.
EXPECT_EQ(0, acm_a_->SetVAD(true, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_TRUE(dtx);
EXPECT_TRUE(vad);
char codec_pcma_temp[] = "PCMA";
RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2, pcma_pltype_);
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
if (test_mode_ != 0) {
@ -366,19 +366,19 @@ void TestStereo::Perform() {
// Test that VAD/DTX cannot be turned on while sending stereo.
EXPECT_EQ(-1, acm_a_->SetVAD(true, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
EXPECT_EQ(-1, acm_a_->SetVAD(true, false, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
EXPECT_EQ(-1, acm_a_->SetVAD(false, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_FALSE(dtx);
EXPECT_FALSE(vad);
@ -603,7 +603,7 @@ void TestStereo::Perform() {
// Make sure it is possible to set VAD/CNG, now that we are sending mono
// again.
EXPECT_EQ(0, acm_a_->SetVAD(true, true, VADNormal));
EXPECT_EQ(0, acm_a_->VAD(dtx, vad, vad_mode));
EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
EXPECT_TRUE(dtx);
EXPECT_TRUE(vad);
EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
@ -687,7 +687,7 @@ void TestStereo::Perform() {
opus_pltype_);
CodecInst opus_codec_param;
for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, opus_codec_param));
EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
if (!strcmp(opus_codec_param.plname, "opus")) {
opus_codec_param.channels = 1;
EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
@ -821,7 +821,7 @@ void TestStereo::RegisterSendCodec(char side, char* codec_name,
CodecInst my_codec_param;
// Get all codec parameters before registering
CHECK_ERROR(AudioCodingModule::Codec(codec_name, my_codec_param,
CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
sampling_freq_hz, channels));
my_codec_param.rate = rate;
my_codec_param.pacsize = pack_size;
@ -888,7 +888,7 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
}
// Run received side of ACM
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz_b, audio_frame));
CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
// Write output speech to file
out_file_.Write10MsData(
@ -919,11 +919,11 @@ void TestStereo::OpenOutFile(WebRtc_Word16 test_number) {
void TestStereo::DisplaySendReceiveCodec() {
CodecInst my_codec_param;
acm_a_->SendCodec(my_codec_param);
acm_a_->SendCodec(&my_codec_param);
if (test_mode_ != 0) {
printf("%s -> ", my_codec_param.plname);
}
acm_b_->ReceiveCodec(my_codec_param);
acm_b_->ReceiveCodec(&my_codec_param);
if (test_mode_ != 0) {
printf("%s\n", my_codec_param.plname);
}

16
webrtc/modules/audio_coding/main/test/TestVADDTX.cc
View File

@ -78,7 +78,7 @@ void TestVADDTX::Perform()
}
for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
{
_acmB->Codec(n, myCodecParam);
_acmB->Codec(n, &myCodecParam);
if(_testMode != 0)
{
printf("%s\n", myCodecParam.plname);
@ -174,7 +174,7 @@ void TestVADDTX::runTestCases()
if(_testMode != 0)
{
CodecInst myCodecParam;
_acmA->SendCodec(myCodecParam);
_acmA->SendCodec(&myCodecParam);
printf("%s\n", myCodecParam.plname);
}
else
@ -239,7 +239,7 @@ void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
if (_acmA->SetVAD(statusDTX, statusVAD, (ACMVADMode) vadMode) < 0) {
assert(false);
}
if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
if (_acmA->VAD(&dtxEnabled, &vadEnabled, &vadModeSet) < 0) {
assert(false);
}
@ -282,7 +282,7 @@ VADDTXstruct TestVADDTX::GetVAD()
bool dtxEnabled, vadEnabled;
ACMVADMode vadModeSet;
if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
if (_acmA->VAD(&dtxEnabled, &vadEnabled, &vadModeSet) < 0) {
assert(false);
}
@ -328,7 +328,7 @@ WebRtc_Word16 TestVADDTX::RegisterSendCodec(char side,
for(WebRtc_Word16 codecCntr = 0; codecCntr < myACM->NumberOfCodecs();
codecCntr++)
{
CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, myCodecParam));
CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, &myCodecParam));
if(!STR_CASE_CMP(myCodecParam.plname, codecName))
{
if((samplingFreqHz == -1) || (myCodecParam.plfreq == samplingFreqHz))
@ -366,7 +366,7 @@ void TestVADDTX::Run()
CHECK_ERROR(_acmA->Process());
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
}
#ifdef PRINT_STAT
@ -399,7 +399,7 @@ WebRtc_Word16 TestVADDTX::VerifyTest()
WebRtc_UWord8 vadPattern = 0;
WebRtc_UWord8 emptyFramePattern[6];
CodecInst myCodecParam;
_acmA->SendCodec(myCodecParam);
_acmA->SendCodec(&myCodecParam);
bool dtxInUse = true;
bool isReplaced = false;
if ((STR_CASE_CMP(myCodecParam.plname,"G729") == 0) ||
@ -408,7 +408,7 @@ WebRtc_Word16 TestVADDTX::VerifyTest()
(STR_CASE_CMP(myCodecParam.plname,"AMR-wb") == 0) ||
(STR_CASE_CMP(myCodecParam.plname,"speex") == 0))
{
_acmA->IsInternalDTXReplacedWithWebRtc(isReplaced);
_acmA->IsInternalDTXReplacedWithWebRtc(&isReplaced);
if (!isReplaced)
{
dtxInUse = false;

26
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
View File

@ -78,7 +78,7 @@ TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A,
printf("========================\n");
for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
{
tmpACM->Codec(codecCntr, codecInst);
tmpACM->Codec(codecCntr, &codecInst);
printf("%d- %s\n", codecCntr, codecInst.plname);
}
printf("\nChoose a send codec for side A [0]: ");
@ -110,10 +110,10 @@ WebRtc_Word16 TwoWayCommunication::SetUp()
CodecInst codecInst_A;
CodecInst codecInst_B;
CodecInst dummyCodec;
_acmA->Codec(codecID_A, codecInst_A);
_acmB->Codec(codecID_B, codecInst_B);
_acmA->Codec(codecID_A, &codecInst_A);
_acmB->Codec(codecID_B, &codecInst_B);
_acmA->Codec(6, dummyCodec);
_acmA->Codec(6, &dummyCodec);
//--- Set A codecs
CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
@ -214,9 +214,9 @@ WebRtc_Word16 TwoWayCommunication::SetUpAutotest()
CodecInst codecInst_B;
CodecInst dummyCodec;
_acmA->Codec("ISAC", codecInst_A, 16000, 1);
_acmB->Codec("L16", codecInst_B, 8000, 1);
_acmA->Codec(6, dummyCodec);
_acmA->Codec("ISAC", &codecInst_A, 16000, 1);
_acmB->Codec("L16", &codecInst_B, 8000, 1);
_acmA->Codec(6, &dummyCodec);
//--- Set A codecs
CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
@ -320,7 +320,7 @@ TwoWayCommunication::Perform()
CodecInst codecInst_B;
CodecInst dummy;
_acmB->SendCodec(codecInst_B);
_acmB->SendCodec(&codecInst_B);
if(_testMode != 0)
{
@ -345,16 +345,16 @@ TwoWayCommunication::Perform()
_acmRefA->Process();
_acmRefB->Process();
_acmA->PlayoutData10Ms(outFreqHzA, audioFrame);
_acmA->PlayoutData10Ms(outFreqHzA, &audioFrame);
_outFileA.Write10MsData(audioFrame);
_acmRefA->PlayoutData10Ms(outFreqHzA, audioFrame);
_acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame);
_outFileRefA.Write10MsData(audioFrame);
_acmB->PlayoutData10Ms(outFreqHzB, audioFrame);
_acmB->PlayoutData10Ms(outFreqHzB, &audioFrame);
_outFileB.Write10MsData(audioFrame);
_acmRefB->PlayoutData10Ms(outFreqHzB, audioFrame);
_acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame);
_outFileRefB.Write10MsData(audioFrame);
msecPassed += 10;
@ -398,7 +398,7 @@ TwoWayCommunication::Perform()
printf("Register Send Codec (audio back in side A)\n");
}
CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
CHECK_ERROR(_acmB->SendCodec(dummy));
CHECK_ERROR(_acmB->SendCodec(&dummy));
}
if(((secPassed%7) == 6) && (msecPassed == 0))
{

10
webrtc/modules/audio_coding/main/test/delay_test.cc
View File

@ -108,7 +108,7 @@ class DelayTest {
WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for(int n = 0; n < num_encoders; n++) {
acm_b_->Codec(n, my_codec_param);
acm_b_->Codec(n, &my_codec_param);
if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
my_codec_param.channels = 1;
else if (my_codec_param.channels > 1)
@ -155,7 +155,7 @@ class DelayTest {
void SendCodec(const CodecConfig& config) {
CodecInst my_codec_param;
ASSERT_EQ(0, AudioCodingModule::Codec(config.name, my_codec_param,
ASSERT_EQ(0, AudioCodingModule::Codec(config.name, &my_codec_param,
config.sample_rate_hz,
config.num_channels));
encoding_sample_rate_hz_ = my_codec_param.plfreq;
@ -201,7 +201,7 @@ class DelayTest {
// Print delay information every 16 frame
if ((num_frames & 0x3F) == 0x3F) {
ACMNetworkStatistics statistics;
acm_b_->NetworkStatistics(statistics);
acm_b_->NetworkStatistics(&statistics);
fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
" ts-based average = %6.3f, "
"curr buff-lev = %4u opt buff-lev = %4u \n",
@ -218,11 +218,11 @@ class DelayTest {
in_file_a_.Read10MsData(audio_frame);
ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, audio_frame));
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
out_file_b_.Write10MsData(audio_frame.data_,
audio_frame.samples_per_channel_ *
audio_frame.num_channels_);
acm_b_->PlayoutTimestamp(playout_ts);
acm_b_->PlayoutTimestamp(&playout_ts);
received_ts = channel_a2b_->LastInTimestamp();
inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts) /
static_cast<double>(encoding_sample_rate_hz_);

8
webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
View File

@ -111,7 +111,7 @@ void DualStreamTest::PopulateCodecInstances(int frame_size_primary_ms,
red_encoder_.pltype = -1;
for (int n = 0; n < AudioCodingModule::NumberOfCodecs(); n++) {
AudioCodingModule::Codec(n, my_codec);
AudioCodingModule::Codec(n, &my_codec);
if (strcmp(my_codec.plname, "ISAC") == 0 &&
my_codec.plfreq == sampling_rate) {
my_codec.rate = 32000;
@ -480,7 +480,7 @@ TEST_F(DualStreamTest, Api) {
bool vad_status;
bool dtx_status;
ACMVADMode vad_mode;
EXPECT_EQ(0, acm_dual_stream_->VAD(vad_status, dtx_status, vad_mode));
EXPECT_EQ(0, acm_dual_stream_->VAD(&vad_status, &dtx_status, &vad_mode));
EXPECT_TRUE(vad_status);
EXPECT_TRUE(dtx_status);
EXPECT_EQ(VADNormal, vad_mode);
@ -492,7 +492,7 @@ TEST_F(DualStreamTest, Api) {
ASSERT_EQ(0, memcmp(&my_codec, &secondary_encoder_, sizeof(my_codec)));
// Test if VAD get disabled after registering secondary codec.
EXPECT_EQ(0, acm_dual_stream_->VAD(vad_status, dtx_status, vad_mode));
EXPECT_EQ(0, acm_dual_stream_->VAD(&vad_status, &dtx_status, &vad_mode));
EXPECT_FALSE(vad_status);
EXPECT_FALSE(dtx_status);
@ -506,7 +506,7 @@ TEST_F(DualStreamTest, Api) {
ASSERT_EQ(0, acm_dual_stream_->SetVAD(true, true, VADVeryAggr));
// Make sure VAD is activated.
EXPECT_EQ(0, acm_dual_stream_->VAD(vad_status, dtx_status, vad_mode));
EXPECT_EQ(0, acm_dual_stream_->VAD(&vad_status, &dtx_status, &vad_mode));
EXPECT_TRUE(vad_status);
EXPECT_TRUE(dtx_status);
EXPECT_EQ(VADVeryAggr, vad_mode);

16
webrtc/modules/audio_coding/main/test/iSACTest.cc
View File

@ -55,7 +55,7 @@ WebRtc_Word16 SetISAConfig(
(isacConfig.currentFrameSizeMsec != 0))
{
CodecInst sendCodec;
acm->SendCodec(sendCodec);
acm->SendCodec(&sendCodec);
if(isacConfig.currentRateBitPerSec < 0)
{
sendCodec.rate = -1;
@ -155,7 +155,7 @@ ISACTest::Setup()
for(codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs(); codecCntr++)
{
AudioCodingModule::Codec(codecCntr, codecParam);
AudioCodingModule::Codec(codecCntr, &codecParam);
if(!STR_CASE_CMP(codecParam.plname, "ISAC") && codecParam.plfreq == 16000)
{
memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
@ -210,14 +210,14 @@ ISACTest::Setup()
Run10ms();
}
CodecInst receiveCodec;
CHECK_ERROR(_acmA->ReceiveCodec(receiveCodec));
CHECK_ERROR(_acmA->ReceiveCodec(&receiveCodec));
if(_testMode != 0)
{
printf("Side A Receive Codec\n");
printf("%s %d\n", receiveCodec.plname, receiveCodec.plfreq);
}
CHECK_ERROR(_acmB->ReceiveCodec(receiveCodec));
CHECK_ERROR(_acmB->ReceiveCodec(&receiveCodec));
if(_testMode != 0)
{
printf("Side B Receive Codec\n");
@ -357,10 +357,10 @@ ISACTest::Run10ms()
CHECK_ERROR(_acmA->Process());
CHECK_ERROR(_acmB->Process());
CHECK_ERROR(_acmA->PlayoutData10Ms(32000, audioFrame));
CHECK_ERROR(_acmA->PlayoutData10Ms(32000, &audioFrame));
_outFileA.Write10MsData(audioFrame);
CHECK_ERROR(_acmB->PlayoutData10Ms(32000, audioFrame));
CHECK_ERROR(_acmB->PlayoutData10Ms(32000, &audioFrame));
_outFileB.Write10MsData(audioFrame);
}
@ -444,9 +444,9 @@ ISACTest::EncodeDecode(
{
myEvent->Wait(5000);
_acmA->SendCodec(sendCodec);
_acmA->SendCodec(&sendCodec);
if(_testMode == 2) printf("[%d] ", sendCodec.rate);
_acmB->SendCodec(sendCodec);
_acmB->SendCodec(&sendCodec);
if(_testMode == 2) printf("[%d] ", sendCodec.rate);
}
}

16
webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
View File

@ -80,7 +80,7 @@ class InitialPlayoutDelayTest : public ::testing::Test {
const int kChannels[2] = {1, 2};
for (int n = 0; n < 3; ++n) {
for (int k = 0; k < 2; ++k) {
AudioCodingModule::Codec("L16", codec, kFsHz[n], kChannels[k]);
AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
acm_b_->RegisterReceiveCodec(codec);
}
}
@ -114,7 +114,7 @@ class InitialPlayoutDelayTest : public ::testing::Test {
timestamp += in_audio_frame.samples_per_channel_;
ASSERT_EQ(0, acm_a_->Add10MsData(in_audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, out_audio_frame));
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
rms = FrameRms(out_audio_frame);
++num_frames;
}
@ -131,38 +131,38 @@ class InitialPlayoutDelayTest : public ::testing::Test {
TEST_F( InitialPlayoutDelayTest, NbMono) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 8000, 1);
AudioCodingModule::Codec("L16", &codec, 8000, 1);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, WbMono) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 16000, 1);
AudioCodingModule::Codec("L16", &codec, 16000, 1);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, SwbMono) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 32000, 1);
AudioCodingModule::Codec("L16", &codec, 32000, 1);
Run(codec, 2000); // NetEq buffer is not sufficiently large for 3 sec of
// PCM16 super-wideband.
}
TEST_F( InitialPlayoutDelayTest, NbStereo) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 8000, 2);
AudioCodingModule::Codec("L16", &codec, 8000, 2);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, WbStereo) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 16000, 2);
AudioCodingModule::Codec("L16", &codec, 16000, 2);
Run(codec, 3000);
}
TEST_F( InitialPlayoutDelayTest, SwbStereo) {
CodecInst codec;
AudioCodingModule::Codec("L16", codec, 32000, 2);
AudioCodingModule::Codec("L16", &codec, 32000, 2);
Run(codec, 2000); // NetEq buffer is not sufficiently large for 3 sec of
// PCM16 super-wideband.
}

4
webrtc/modules/audio_coding/main/test/utility.cc
View File

@ -138,7 +138,7 @@ ChooseCodec(
}
} while(outOfRange);
CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, codecInst));
CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, &codecInst));
return 0;
}
@ -151,7 +151,7 @@ PrintCodecs()
printf("No Name [Hz] [bps]\n");
for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
{
AudioCodingModule::Codec(codecCntr, codecInst);
AudioCodingModule::Codec(codecCntr, &codecInst);
printf("%2d- %-18s %5d %6d\n",
codecCntr, codecInst.plname, codecInst.plfreq, codecInst.rate);
}