Revert "Activate ACM test for Android in modules_tests." (rev5364).
TBR=turaj@webrtc.org,tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6999006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
parent
44461fa5cb
commit
d0b436a935
@ -63,11 +63,11 @@ void TestFEC::Perform() {
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return;
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#endif
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char nameG722[] = "G722";
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RegisterSendCodec('A', nameG722, 16000);
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EXPECT_EQ(0, RegisterSendCodec('A', nameG722, 16000));
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char nameCN[] = "CN";
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RegisterSendCodec('A', nameCN, 16000);
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EXPECT_EQ(0, RegisterSendCodec('A', nameCN, 16000));
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char nameRED[] = "RED";
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RegisterSendCodec('A', nameRED);
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EXPECT_EQ(0, RegisterSendCodec('A', nameRED));
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OpenOutFile(_testCntr);
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EXPECT_EQ(0, SetVAD(true, true, VADAggr));
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EXPECT_EQ(0, _acmA->SetFECStatus(false));
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@ -81,9 +81,6 @@ void TestFEC::Perform() {
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Run();
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_outFileB.Close();
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// FEC for iSAC is different that other codecs, therefore, we expect that iSAC
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// be enabled for this test. The following is common for both floating-point
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// and fixed-point implementations.
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char nameISAC[] = "iSAC";
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RegisterSendCodec('A', nameISAC, 16000);
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OpenOutFile(_testCntr);
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@ -99,8 +96,6 @@ void TestFEC::Perform() {
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Run();
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_outFileB.Close();
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#if (defined(WEBRTC_CODEC_ISAC))
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// Only for floating-point implementation, where super-wideband is supported.
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RegisterSendCodec('A', nameISAC, 32000);
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OpenOutFile(_testCntr);
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EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
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@ -134,26 +129,11 @@ void TestFEC::Perform() {
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EXPECT_TRUE(_acmA->FECStatus());
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Run();
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_outFileB.Close();
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#else
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// For fixed-point implementation.
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OpenOutFile(_testCntr);
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EXPECT_EQ(0, SetVAD(false, false, VADVeryAggr));
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EXPECT_EQ(0, _acmA->SetFECStatus(false));
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EXPECT_FALSE(_acmA->FECStatus());
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Run();
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_outFileB.Close();
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EXPECT_EQ(0, _acmA->SetFECStatus(true));
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EXPECT_TRUE(_acmA->FECStatus());
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OpenOutFile(_testCntr);
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Run();
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_outFileB.Close();
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#endif
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_channelA2B->SetFECTestWithPacketLoss(true);
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RegisterSendCodec('A', nameG722);
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RegisterSendCodec('A', nameCN, 16000);
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EXPECT_EQ(0, RegisterSendCodec('A', nameG722));
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EXPECT_EQ(0, RegisterSendCodec('A', nameCN, 16000));
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OpenOutFile(_testCntr);
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EXPECT_EQ(0, SetVAD(true, true, VADAggr));
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EXPECT_EQ(0, _acmA->SetFECStatus(false));
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@ -181,8 +161,6 @@ void TestFEC::Perform() {
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Run();
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_outFileB.Close();
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#if (defined(WEBRTC_CODEC_ISAC))
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// Only for floating-point implementation, where super-wideband is supported.
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RegisterSendCodec('A', nameISAC, 32000);
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OpenOutFile(_testCntr);
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EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
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@ -216,31 +194,16 @@ void TestFEC::Perform() {
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EXPECT_TRUE(_acmA->FECStatus());
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Run();
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_outFileB.Close();
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#else
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// For fixed-point implementation.
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OpenOutFile(_testCntr);
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EXPECT_EQ(0, SetVAD(false, false, VADVeryAggr));
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EXPECT_EQ(0, _acmA->SetFECStatus(false));
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EXPECT_FALSE(_acmA->FECStatus());
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Run();
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_outFileB.Close();
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EXPECT_EQ(0, _acmA->SetFECStatus(true));
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EXPECT_TRUE(_acmA->FECStatus());
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OpenOutFile(_testCntr);
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Run();
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_outFileB.Close();
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#endif
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}
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int32_t TestFEC::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {
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return _acmA->SetVAD(enableDTX, enableVAD, vadMode);
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}
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void TestFEC::RegisterSendCodec(char side, char* codecName,
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int16_t TestFEC::RegisterSendCodec(char side, char* codecName,
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int32_t samplingFreqHz) {
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std::cout << std::flush;
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AudioCodingModule* myACM = NULL;
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AudioCodingModule* myACM;
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switch (side) {
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case 'A': {
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myACM = _acmA.get();
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@ -251,15 +214,20 @@ void TestFEC::RegisterSendCodec(char side, char* codecName,
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break;
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}
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default:
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ASSERT_TRUE(false);
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return -1;
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}
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ASSERT_TRUE(myACM != NULL);
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if (myACM == NULL) {
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assert(false);
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return -1;
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}
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CodecInst myCodecParam;
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ASSERT_GT(AudioCodingModule::Codec(codecName, &myCodecParam,
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EXPECT_GT(AudioCodingModule::Codec(codecName, &myCodecParam,
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samplingFreqHz, 1), -1);
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ASSERT_GT(myACM->RegisterSendCodec(myCodecParam), -1);
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EXPECT_GT(myACM->RegisterSendCodec(myCodecParam), -1);
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// Initialization was successful.
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return 0;
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}
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void TestFEC::Run() {
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@ -30,8 +30,8 @@ class TestFEC : public ACMTest {
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// The default value of '-1' indicates that the registration is based only on
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// codec name and a sampling frequency matching is not required. This is
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// useful for codecs which support several sampling frequency.
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void RegisterSendCodec(char side, char* codecName,
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int32_t sampFreqHz = -1);
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int16_t RegisterSendCodec(char side, char* codecName,
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int32_t sampFreqHz = -1);
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void Run();
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void OpenOutFile(int16_t testNumber);
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int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
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@ -809,14 +809,7 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
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channel->reset_payload_size();
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int error_count = 0;
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#ifdef WEBRTC_ARCH_ARM
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const int kMaxNumProcessedFrames = 100; // Limit to 1 second of audio.
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#else
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const int kMaxNumProcessedFrames = 3000; // Limit to 30 second of audio.
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#endif
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int num_frames = 0;
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while (num_frames < kMaxNumProcessedFrames) {
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while (1) {
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// Simulate packet loss by setting |packet_loss_| to "true" in
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// |percent_loss| percent of the loops.
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if (percent_loss > 0) {
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@ -870,15 +863,16 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
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out_file_.Write10MsData(
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audio_frame.data_,
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audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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++num_frames;
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}
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EXPECT_EQ(0, error_count);
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in_file_mono_->Rewind();
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in_file_stereo_->Rewind();
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if (in_file_mono_->EndOfFile()) {
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in_file_mono_->Rewind();
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}
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if (in_file_stereo_->EndOfFile()) {
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in_file_stereo_->Rewind();
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}
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// Reset in case we ended with a lost packet
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channel->set_lost_packet(false);
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}
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@ -50,7 +50,7 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
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Trace::ReturnTrace();
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}
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TEST(AudioCodingModuleTest, TestEncodeDecode) {
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TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
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Trace::CreateTrace();
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Trace::SetTraceFile((webrtc::test::OutputPath() +
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"acm_encodedecode_trace.txt").c_str());
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@ -65,7 +65,7 @@ TEST(AudioCodingModuleTest, TestEncodeDecode) {
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Trace::ReturnTrace();
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}
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TEST(AudioCodingModuleTest, TestFEC) {
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TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestFEC)) {
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Trace::CreateTrace();
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Trace::SetTraceFile((webrtc::test::OutputPath() +
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"acm_fec_trace.txt").c_str());
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@ -80,7 +80,7 @@ TEST(AudioCodingModuleTest, TestFEC) {
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Trace::ReturnTrace();
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}
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TEST(AudioCodingModuleTest, TestIsac) {
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TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) {
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Trace::CreateTrace();
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Trace::SetTraceFile((webrtc::test::OutputPath() +
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"acm_isac_trace.txt").c_str());
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@ -95,7 +95,7 @@ TEST(AudioCodingModuleTest, TestIsac) {
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Trace::ReturnTrace();
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}
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TEST(AudioCodingModuleTest, TwoWayCommunication) {
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TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunication)) {
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Trace::CreateTrace();
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Trace::SetTraceFile((webrtc::test::OutputPath() +
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"acm_twowaycom_trace.txt").c_str());
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@ -110,7 +110,7 @@ TEST(AudioCodingModuleTest, TwoWayCommunication) {
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Trace::ReturnTrace();
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}
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TEST(AudioCodingModuleTest, TestStereo) {
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TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
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Trace::CreateTrace();
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Trace::SetTraceFile((webrtc::test::OutputPath() +
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"acm_stereo_trace.txt").c_str());
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@ -125,7 +125,7 @@ TEST(AudioCodingModuleTest, TestStereo) {
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Trace::ReturnTrace();
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}
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TEST(AudioCodingModuleTest, TestVADDTX) {
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TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestVADDTX)) {
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Trace::CreateTrace();
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Trace::SetTraceFile((webrtc::test::OutputPath() +
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"acm_vaddtx_trace.txt").c_str());
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@ -94,24 +94,6 @@ ISACTest::ISACTest(int testMode, const Config& config)
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ISACTest::~ISACTest() {}
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void ISACTest::Run10ms() {
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AudioFrame audioFrame;
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EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
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EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
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EXPECT_EQ(0, _acmB->Add10MsData(audioFrame));
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EXPECT_GT(_acmA->Process(), -1);
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EXPECT_GT(_acmB->Process(), -1);
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EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
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_outFileA.Write10MsData(audioFrame);
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EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
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_outFileB.Write10MsData(audioFrame);
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}
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#if (defined(WEBRTC_CODEC_ISAC))
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// Depending on whether the floating-point iSAC is activated the following
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// implementations would differ.
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void ISACTest::Setup() {
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int codecCntr;
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CodecInst codecParam;
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@ -262,6 +244,19 @@ void ISACTest::Perform() {
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}
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}
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void ISACTest::Run10ms() {
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AudioFrame audioFrame;
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EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
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EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
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EXPECT_EQ(0, _acmB->Add10MsData(audioFrame));
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EXPECT_GT(_acmA->Process(), -1);
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EXPECT_GT(_acmB->Process(), -1);
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EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
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_outFileA.Write10MsData(audioFrame);
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EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
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_outFileB.Write10MsData(audioFrame);
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}
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void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
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ACMTestISACConfig& swbISACConfig) {
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// Files in Side A and B
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@ -322,6 +317,9 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
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_channel_B2A->PrintStats(_paramISAC16kHz);
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}
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_channel_A2B->ResetStats();
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_channel_B2A->ResetStats();
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_outFileA.Close();
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_outFileB.Close();
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_inFileA.Close();
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@ -394,210 +392,5 @@ void ISACTest::SwitchingSamplingRate(int testNr, int maxSampRateChange) {
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_inFileA.Close();
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_inFileB.Close();
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}
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#else // Only iSAC fixed-point is defined.
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static int PayloadSizeToInstantaneousRate(int payload_size_bytes,
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int frame_size_ms) {
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return payload_size_bytes * 8 / frame_size_ms / 1000;
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}
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void ISACTest::Setup() {
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CodecInst codec_param;
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codec_param.plfreq = 0; // Invalid value.
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for (int n = 0; n < AudioCodingModule::NumberOfCodecs(); ++n) {
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EXPECT_EQ(0, AudioCodingModule::Codec(n, &codec_param));
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if (!STR_CASE_CMP(codec_param.plname, "ISAC")) {
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ASSERT_EQ(16000, codec_param.plfreq);
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memcpy(&_paramISAC16kHz, &codec_param, sizeof(codec_param));
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_idISAC16kHz = n;
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break;
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}
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}
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EXPECT_GT(codec_param.plfreq, 0);
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EXPECT_EQ(0, _acmA->RegisterReceiveCodec(_paramISAC16kHz));
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EXPECT_EQ(0, _acmB->RegisterReceiveCodec(_paramISAC16kHz));
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//--- Set A-to-B channel
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_channel_A2B.reset(new Channel);
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EXPECT_EQ(0, _acmA->RegisterTransportCallback(_channel_A2B.get()));
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_channel_A2B->RegisterReceiverACM(_acmB.get());
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//--- Set B-to-A channel
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_channel_B2A.reset(new Channel);
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EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get()));
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_channel_B2A->RegisterReceiverACM(_acmA.get());
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file_name_swb_ = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
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"pcm");
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EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
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EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC16kHz));
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}
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void ISACTest::EncodeDecode(int test_number, ACMTestISACConfig& isac_config_a,
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ACMTestISACConfig& isac_config_b) {
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// Files in Side A and B
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_inFileA.Open(file_name_swb_, 32000, "rb", true);
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_inFileB.Open(file_name_swb_, 32000, "rb", true);
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std::string file_name_out;
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std::stringstream file_stream_a;
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std::stringstream file_stream_b;
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file_stream_a << webrtc::test::OutputPath();
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file_stream_b << webrtc::test::OutputPath();
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file_stream_a << "out_iSACTest_A_" << test_number << ".pcm";
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file_stream_b << "out_iSACTest_B_" << test_number << ".pcm";
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file_name_out = file_stream_a.str();
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_outFileA.Open(file_name_out, 32000, "wb");
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file_name_out = file_stream_b.str();
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_outFileB.Open(file_name_out, 32000, "wb");
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CodecInst codec;
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EXPECT_EQ(0, _acmA->SendCodec(&codec));
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EXPECT_EQ(0, _acmB->SendCodec(&codec));
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// Set the configurations.
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SetISAConfig(isac_config_a, _acmA.get(), _testMode);
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SetISAConfig(isac_config_b, _acmB.get(), _testMode);
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bool adaptiveMode = false;
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if (isac_config_a.currentRateBitPerSec == -1 ||
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isac_config_b.currentRateBitPerSec == -1) {
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adaptiveMode = true;
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}
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_channel_A2B->ResetStats();
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_channel_B2A->ResetStats();
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EventWrapper* myEvent = EventWrapper::Create();
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EXPECT_TRUE(myEvent->StartTimer(true, 10));
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while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
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Run10ms();
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if (adaptiveMode && _testMode != 0) {
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myEvent->Wait(5000);
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}
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}
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if (_testMode != 0) {
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printf("\n\nSide A statistics\n\n");
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_channel_A2B->PrintStats(_paramISAC32kHz);
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printf("\n\nSide B statistics\n\n");
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_channel_B2A->PrintStats(_paramISAC16kHz);
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}
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_outFileA.Close();
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_outFileB.Close();
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_inFileA.Close();
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_inFileB.Close();
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}
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void ISACTest::Perform() {
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Setup();
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int16_t test_number = 0;
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ACMTestISACConfig isac_config_a;
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ACMTestISACConfig isac_config_b;
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SetISACConfigDefault(isac_config_a);
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SetISACConfigDefault(isac_config_b);
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// Instantaneous mode.
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isac_config_a.currentRateBitPerSec = 32000;
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isac_config_b.currentRateBitPerSec = 12000;
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EncodeDecode(test_number, isac_config_a, isac_config_b);
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test_number++;
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SetISACConfigDefault(isac_config_a);
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SetISACConfigDefault(isac_config_b);
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// Channel adaptive.
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isac_config_a.currentRateBitPerSec = -1;
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isac_config_b.currentRateBitPerSec = -1;
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isac_config_a.initRateBitPerSec = 13000;
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isac_config_a.initFrameSizeInMsec = 60;
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isac_config_a.enforceFrameSize = true;
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isac_config_a.currentFrameSizeMsec = 60;
|
||||
isac_config_b.initRateBitPerSec = 20000;
|
||||
isac_config_b.initFrameSizeInMsec = 30;
|
||||
EncodeDecode(test_number, isac_config_a, isac_config_b);
|
||||
test_number++;
|
||||
|
||||
SetISACConfigDefault(isac_config_a);
|
||||
SetISACConfigDefault(isac_config_b);
|
||||
isac_config_a.currentRateBitPerSec = 32000;
|
||||
isac_config_b.currentRateBitPerSec = 32000;
|
||||
isac_config_a.currentFrameSizeMsec = 30;
|
||||
isac_config_b.currentFrameSizeMsec = 60;
|
||||
|
||||
int user_input;
|
||||
const int kMaxPayloadLenBytes30MSec = 110;
|
||||
const int kMaxPayloadLenBytes60MSec = 160;
|
||||
if ((_testMode == 0) || (_testMode == 1)) {
|
||||
isac_config_a.maxPayloadSizeByte =
|
||||
static_cast<uint16_t>(kMaxPayloadLenBytes30MSec);
|
||||
isac_config_b.maxPayloadSizeByte =
|
||||
static_cast<uint16_t>(kMaxPayloadLenBytes60MSec);
|
||||
} else {
|
||||
printf("Enter the max payload-size for side A: ");
|
||||
CHECK_ERROR(scanf("%d", &user_input));
|
||||
isac_config_a.maxPayloadSizeByte = (uint16_t) user_input;
|
||||
printf("Enter the max payload-size for side B: ");
|
||||
CHECK_ERROR(scanf("%d", &user_input));
|
||||
isac_config_b.maxPayloadSizeByte = (uint16_t) user_input;
|
||||
}
|
||||
EncodeDecode(test_number, isac_config_a, isac_config_b);
|
||||
test_number++;
|
||||
|
||||
ACMTestPayloadStats payload_stats;
|
||||
_channel_A2B->Stats(_paramISAC16kHz, payload_stats);
|
||||
EXPECT_GT(payload_stats.frameSizeStats[0].maxPayloadLen, 0);
|
||||
EXPECT_LE(payload_stats.frameSizeStats[0].maxPayloadLen,
|
||||
static_cast<int>(isac_config_a.maxPayloadSizeByte));
|
||||
_channel_B2A->Stats(_paramISAC16kHz, payload_stats);
|
||||
EXPECT_GT(payload_stats.frameSizeStats[0].maxPayloadLen, 0);
|
||||
EXPECT_LE(payload_stats.frameSizeStats[0].maxPayloadLen,
|
||||
static_cast<int>(isac_config_b.maxPayloadSizeByte));
|
||||
|
||||
_acmA->ResetEncoder();
|
||||
_acmB->ResetEncoder();
|
||||
SetISACConfigDefault(isac_config_a);
|
||||
SetISACConfigDefault(isac_config_b);
|
||||
isac_config_a.currentRateBitPerSec = 32000;
|
||||
isac_config_b.currentRateBitPerSec = 32000;
|
||||
isac_config_a.currentFrameSizeMsec = 30;
|
||||
isac_config_b.currentFrameSizeMsec = 60;
|
||||
|
||||
const int kMaxEncodingRateBitsPerSec = 32000;
|
||||
if ((_testMode == 0) || (_testMode == 1)) {
|
||||
isac_config_a.maxRateBitPerSec =
|
||||
static_cast<uint32_t>(kMaxEncodingRateBitsPerSec);
|
||||
isac_config_b.maxRateBitPerSec =
|
||||
static_cast<uint32_t>(kMaxEncodingRateBitsPerSec);
|
||||
} else {
|
||||
printf("Enter the max rate for side A: ");
|
||||
CHECK_ERROR(scanf("%d", &user_input));
|
||||
isac_config_a.maxRateBitPerSec = (uint32_t) user_input;
|
||||
printf("Enter the max rate for side B: ");
|
||||
CHECK_ERROR(scanf("%d", &user_input));
|
||||
isac_config_b.maxRateBitPerSec = (uint32_t) user_input;
|
||||
}
|
||||
EncodeDecode(test_number, isac_config_a, isac_config_b);
|
||||
|
||||
_channel_A2B->Stats(_paramISAC16kHz, payload_stats);
|
||||
EXPECT_GT(payload_stats.frameSizeStats[0].maxPayloadLen, 0);
|
||||
EXPECT_LE(PayloadSizeToInstantaneousRate(
|
||||
payload_stats.frameSizeStats[0].maxPayloadLen,
|
||||
isac_config_a.currentFrameSizeMsec),
|
||||
static_cast<int>(isac_config_a.maxRateBitPerSec));
|
||||
|
||||
_channel_B2A->Stats(_paramISAC16kHz, payload_stats);
|
||||
EXPECT_GT(payload_stats.frameSizeStats[0].maxPayloadLen, 0);
|
||||
EXPECT_LE(PayloadSizeToInstantaneousRate(
|
||||
payload_stats.frameSizeStats[0].maxPayloadLen,
|
||||
isac_config_b.currentFrameSizeMsec),
|
||||
static_cast<int>(isac_config_b.maxRateBitPerSec));
|
||||
}
|
||||
#endif // WEBRTC_CODEC_ISAC
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user