Activate ACM test for Android in modules_tests.
TEST=local on Nexus 7.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5364 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/TestFEC.cc b/webrtc/modules/audio_coding/main/test/TestFEC.cc
index 032579c..8a1f27e 100644
--- a/webrtc/modules/audio_coding/main/test/TestFEC.cc
+++ b/webrtc/modules/audio_coding/main/test/TestFEC.cc
@@ -63,11 +63,11 @@
return;
#endif
char nameG722[] = "G722";
- EXPECT_EQ(0, RegisterSendCodec('A', nameG722, 16000));
+ RegisterSendCodec('A', nameG722, 16000);
char nameCN[] = "CN";
- EXPECT_EQ(0, RegisterSendCodec('A', nameCN, 16000));
+ RegisterSendCodec('A', nameCN, 16000);
char nameRED[] = "RED";
- EXPECT_EQ(0, RegisterSendCodec('A', nameRED));
+ RegisterSendCodec('A', nameRED);
OpenOutFile(_testCntr);
EXPECT_EQ(0, SetVAD(true, true, VADAggr));
EXPECT_EQ(0, _acmA->SetFECStatus(false));
@@ -81,6 +81,9 @@
Run();
_outFileB.Close();
+ // FEC for iSAC is different that other codecs, therefore, we expect that iSAC
+ // be enabled for this test. The following is common for both floating-point
+ // and fixed-point implementations.
char nameISAC[] = "iSAC";
RegisterSendCodec('A', nameISAC, 16000);
OpenOutFile(_testCntr);
@@ -96,6 +99,8 @@
Run();
_outFileB.Close();
+#if (defined(WEBRTC_CODEC_ISAC))
+ // Only for floating-point implementation, where super-wideband is supported.
RegisterSendCodec('A', nameISAC, 32000);
OpenOutFile(_testCntr);
EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
@@ -129,11 +134,26 @@
EXPECT_TRUE(_acmA->FECStatus());
Run();
_outFileB.Close();
+#else
+ // For fixed-point implementation.
+ OpenOutFile(_testCntr);
+ EXPECT_EQ(0, SetVAD(false, false, VADVeryAggr));
+ EXPECT_EQ(0, _acmA->SetFECStatus(false));
+ EXPECT_FALSE(_acmA->FECStatus());
+ Run();
+ _outFileB.Close();
+
+ EXPECT_EQ(0, _acmA->SetFECStatus(true));
+ EXPECT_TRUE(_acmA->FECStatus());
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+#endif
_channelA2B->SetFECTestWithPacketLoss(true);
- EXPECT_EQ(0, RegisterSendCodec('A', nameG722));
- EXPECT_EQ(0, RegisterSendCodec('A', nameCN, 16000));
+ RegisterSendCodec('A', nameG722);
+ RegisterSendCodec('A', nameCN, 16000);
OpenOutFile(_testCntr);
EXPECT_EQ(0, SetVAD(true, true, VADAggr));
EXPECT_EQ(0, _acmA->SetFECStatus(false));
@@ -161,6 +181,8 @@
Run();
_outFileB.Close();
+#if (defined(WEBRTC_CODEC_ISAC))
+ // Only for floating-point implementation, where super-wideband is supported.
RegisterSendCodec('A', nameISAC, 32000);
OpenOutFile(_testCntr);
EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
@@ -194,16 +216,31 @@
EXPECT_TRUE(_acmA->FECStatus());
Run();
_outFileB.Close();
+#else
+ // For fixed-point implementation.
+ OpenOutFile(_testCntr);
+ EXPECT_EQ(0, SetVAD(false, false, VADVeryAggr));
+ EXPECT_EQ(0, _acmA->SetFECStatus(false));
+ EXPECT_FALSE(_acmA->FECStatus());
+ Run();
+ _outFileB.Close();
+
+ EXPECT_EQ(0, _acmA->SetFECStatus(true));
+ EXPECT_TRUE(_acmA->FECStatus());
+ OpenOutFile(_testCntr);
+ Run();
+ _outFileB.Close();
+#endif
}
int32_t TestFEC::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {
return _acmA->SetVAD(enableDTX, enableVAD, vadMode);
}
-int16_t TestFEC::RegisterSendCodec(char side, char* codecName,
+void TestFEC::RegisterSendCodec(char side, char* codecName,
int32_t samplingFreqHz) {
std::cout << std::flush;
- AudioCodingModule* myACM;
+ AudioCodingModule* myACM = NULL;
switch (side) {
case 'A': {
myACM = _acmA.get();
@@ -214,20 +251,15 @@
break;
}
default:
- return -1;
+ ASSERT_TRUE(false);
}
- if (myACM == NULL) {
- assert(false);
- return -1;
- }
+ ASSERT_TRUE(myACM != NULL);
+
CodecInst myCodecParam;
- EXPECT_GT(AudioCodingModule::Codec(codecName, &myCodecParam,
+ ASSERT_GT(AudioCodingModule::Codec(codecName, &myCodecParam,
samplingFreqHz, 1), -1);
- EXPECT_GT(myACM->RegisterSendCodec(myCodecParam), -1);
-
- // Initialization was successful.
- return 0;
+ ASSERT_GT(myACM->RegisterSendCodec(myCodecParam), -1);
}
void TestFEC::Run() {
diff --git a/webrtc/modules/audio_coding/main/test/TestFEC.h b/webrtc/modules/audio_coding/main/test/TestFEC.h
index af3cdd7..f61e868 100644
--- a/webrtc/modules/audio_coding/main/test/TestFEC.h
+++ b/webrtc/modules/audio_coding/main/test/TestFEC.h
@@ -30,8 +30,8 @@
// The default value of '-1' indicates that the registration is based only on
// codec name and a sampling frequency matching is not required. This is
// useful for codecs which support several sampling frequency.
- int16_t RegisterSendCodec(char side, char* codecName,
- int32_t sampFreqHz = -1);
+ void RegisterSendCodec(char side, char* codecName,
+ int32_t sampFreqHz = -1);
void Run();
void OpenOutFile(int16_t testNumber);
int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index b26334c..88cf963 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -809,7 +809,14 @@
channel->reset_payload_size();
int error_count = 0;
- while (1) {
+#ifdef WEBRTC_ARCH_ARM
+ const int kMaxNumProcessedFrames = 100; // Limit to 1 second of audio.
+#else
+ const int kMaxNumProcessedFrames = 3000; // Limit to 30 second of audio.
+#endif
+
+ int num_frames = 0;
+ while (num_frames < kMaxNumProcessedFrames) {
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
if (percent_loss > 0) {
@@ -863,16 +870,15 @@
out_file_.Write10MsData(
audio_frame.data_,
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+
+ ++num_frames;
}
EXPECT_EQ(0, error_count);
- if (in_file_mono_->EndOfFile()) {
- in_file_mono_->Rewind();
- }
- if (in_file_stereo_->EndOfFile()) {
- in_file_stereo_->Rewind();
- }
+ in_file_mono_->Rewind();
+ in_file_stereo_->Rewind();
+
// Reset in case we ended with a lost packet
channel->set_lost_packet(false);
}
diff --git a/webrtc/modules/audio_coding/main/test/Tester.cc b/webrtc/modules/audio_coding/main/test/Tester.cc
index 31f7317..e089679 100644
--- a/webrtc/modules/audio_coding/main/test/Tester.cc
+++ b/webrtc/modules/audio_coding/main/test/Tester.cc
@@ -50,7 +50,7 @@
Trace::ReturnTrace();
}
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
+TEST(AudioCodingModuleTest, TestEncodeDecode) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_encodedecode_trace.txt").c_str());
@@ -65,7 +65,7 @@
Trace::ReturnTrace();
}
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestFEC)) {
+TEST(AudioCodingModuleTest, TestFEC) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_fec_trace.txt").c_str());
@@ -80,7 +80,7 @@
Trace::ReturnTrace();
}
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) {
+TEST(AudioCodingModuleTest, TestIsac) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_isac_trace.txt").c_str());
@@ -95,7 +95,7 @@
Trace::ReturnTrace();
}
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunication)) {
+TEST(AudioCodingModuleTest, TwoWayCommunication) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_twowaycom_trace.txt").c_str());
@@ -110,7 +110,7 @@
Trace::ReturnTrace();
}
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
+TEST(AudioCodingModuleTest, TestStereo) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_stereo_trace.txt").c_str());
@@ -125,7 +125,7 @@
Trace::ReturnTrace();
}
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestVADDTX)) {
+TEST(AudioCodingModuleTest, TestVADDTX) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_vaddtx_trace.txt").c_str());
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc
index f7fef4a..eb86a4f 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc
@@ -94,6 +94,24 @@
ISACTest::~ISACTest() {}
+void ISACTest::Run10ms() {
+ AudioFrame audioFrame;
+ EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+ EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
+ EXPECT_EQ(0, _acmB->Add10MsData(audioFrame));
+ EXPECT_GT(_acmA->Process(), -1);
+ EXPECT_GT(_acmB->Process(), -1);
+ EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
+ _outFileA.Write10MsData(audioFrame);
+ EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
+ _outFileB.Write10MsData(audioFrame);
+}
+
+
+#if (defined(WEBRTC_CODEC_ISAC))
+// Depending on whether the floating-point iSAC is activated the following
+// implementations would differ.
+
void ISACTest::Setup() {
int codecCntr;
CodecInst codecParam;
@@ -244,19 +262,6 @@
}
}
-void ISACTest::Run10ms() {
- AudioFrame audioFrame;
- EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmA->Add10MsData(audioFrame));
- EXPECT_EQ(0, _acmB->Add10MsData(audioFrame));
- EXPECT_GT(_acmA->Process(), -1);
- EXPECT_GT(_acmB->Process(), -1);
- EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
- _outFileA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
- _outFileB.Write10MsData(audioFrame);
-}
-
void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig) {
// Files in Side A and B
@@ -317,9 +322,6 @@
_channel_B2A->PrintStats(_paramISAC16kHz);
}
- _channel_A2B->ResetStats();
- _channel_B2A->ResetStats();
-
_outFileA.Close();
_outFileB.Close();
_inFileA.Close();
@@ -392,5 +394,210 @@
_inFileA.Close();
_inFileB.Close();
}
+#else // Only iSAC fixed-point is defined.
+
+static int PayloadSizeToInstantaneousRate(int payload_size_bytes,
+ int frame_size_ms) {
+ return payload_size_bytes * 8 / frame_size_ms / 1000;
+}
+
+void ISACTest::Setup() {
+ CodecInst codec_param;
+ codec_param.plfreq = 0; // Invalid value.
+ for (int n = 0; n < AudioCodingModule::NumberOfCodecs(); ++n) {
+ EXPECT_EQ(0, AudioCodingModule::Codec(n, &codec_param));
+ if (!STR_CASE_CMP(codec_param.plname, "ISAC")) {
+ ASSERT_EQ(16000, codec_param.plfreq);
+ memcpy(&_paramISAC16kHz, &codec_param, sizeof(codec_param));
+ _idISAC16kHz = n;
+ break;
+ }
+ }
+ EXPECT_GT(codec_param.plfreq, 0);
+
+ EXPECT_EQ(0, _acmA->RegisterReceiveCodec(_paramISAC16kHz));
+ EXPECT_EQ(0, _acmB->RegisterReceiveCodec(_paramISAC16kHz));
+
+ //--- Set A-to-B channel
+ _channel_A2B.reset(new Channel);
+ EXPECT_EQ(0, _acmA->RegisterTransportCallback(_channel_A2B.get()));
+ _channel_A2B->RegisterReceiverACM(_acmB.get());
+
+ //--- Set B-to-A channel
+ _channel_B2A.reset(new Channel);
+ EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get()));
+ _channel_B2A->RegisterReceiverACM(_acmA.get());
+
+ file_name_swb_ = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
+ "pcm");
+
+ EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
+ EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC16kHz));
+}
+
+void ISACTest::EncodeDecode(int test_number, ACMTestISACConfig& isac_config_a,
+ ACMTestISACConfig& isac_config_b) {
+ // Files in Side A and B
+ _inFileA.Open(file_name_swb_, 32000, "rb", true);
+ _inFileB.Open(file_name_swb_, 32000, "rb", true);
+
+ std::string file_name_out;
+ std::stringstream file_stream_a;
+ std::stringstream file_stream_b;
+ file_stream_a << webrtc::test::OutputPath();
+ file_stream_b << webrtc::test::OutputPath();
+ file_stream_a << "out_iSACTest_A_" << test_number << ".pcm";
+ file_stream_b << "out_iSACTest_B_" << test_number << ".pcm";
+ file_name_out = file_stream_a.str();
+ _outFileA.Open(file_name_out, 32000, "wb");
+ file_name_out = file_stream_b.str();
+ _outFileB.Open(file_name_out, 32000, "wb");
+
+ CodecInst codec;
+ EXPECT_EQ(0, _acmA->SendCodec(&codec));
+ EXPECT_EQ(0, _acmB->SendCodec(&codec));
+
+ // Set the configurations.
+ SetISAConfig(isac_config_a, _acmA.get(), _testMode);
+ SetISAConfig(isac_config_b, _acmB.get(), _testMode);
+
+ bool adaptiveMode = false;
+ if (isac_config_a.currentRateBitPerSec == -1 ||
+ isac_config_b.currentRateBitPerSec == -1) {
+ adaptiveMode = true;
+ }
+ _channel_A2B->ResetStats();
+ _channel_B2A->ResetStats();
+
+ EventWrapper* myEvent = EventWrapper::Create();
+ EXPECT_TRUE(myEvent->StartTimer(true, 10));
+ while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
+ Run10ms();
+ if (adaptiveMode && _testMode != 0) {
+ myEvent->Wait(5000);
+ }
+ }
+
+ if (_testMode != 0) {
+ printf("\n\nSide A statistics\n\n");
+ _channel_A2B->PrintStats(_paramISAC32kHz);
+
+ printf("\n\nSide B statistics\n\n");
+ _channel_B2A->PrintStats(_paramISAC16kHz);
+ }
+
+ _outFileA.Close();
+ _outFileB.Close();
+ _inFileA.Close();
+ _inFileB.Close();
+}
+
+void ISACTest::Perform() {
+ Setup();
+
+ int16_t test_number = 0;
+ ACMTestISACConfig isac_config_a;
+ ACMTestISACConfig isac_config_b;
+
+ SetISACConfigDefault(isac_config_a);
+ SetISACConfigDefault(isac_config_b);
+
+ // Instantaneous mode.
+ isac_config_a.currentRateBitPerSec = 32000;
+ isac_config_b.currentRateBitPerSec = 12000;
+ EncodeDecode(test_number, isac_config_a, isac_config_b);
+ test_number++;
+
+ SetISACConfigDefault(isac_config_a);
+ SetISACConfigDefault(isac_config_b);
+
+ // Channel adaptive.
+ isac_config_a.currentRateBitPerSec = -1;
+ isac_config_b.currentRateBitPerSec = -1;
+ isac_config_a.initRateBitPerSec = 13000;
+ isac_config_a.initFrameSizeInMsec = 60;
+ isac_config_a.enforceFrameSize = true;
+ 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
