Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded.
The main() was deleted in r4731.
BUG=
R=andrew@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2370004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/main/source/audio_coding_module.gypi b/modules/audio_coding/main/source/audio_coding_module.gypi
index c6c10f6..a0389b0 100644
--- a/modules/audio_coding/main/source/audio_coding_module.gypi
+++ b/modules/audio_coding/main/source/audio_coding_module.gypi
@@ -117,7 +117,7 @@
'dependencies': [
'audio_coding_module',
'<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/test/test.gyp:test_support_main',
+ '<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
@@ -125,6 +125,7 @@
'../test/delay_test.cc',
'../test/Channel.cc',
'../test/PCMFile.cc',
+ '../test/utility.cc',
],
}, # delay_test
{
@@ -133,7 +134,7 @@
'dependencies': [
'audio_coding_module',
'<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/test/test.gyp:test_support_main',
+ '<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
diff --git a/modules/audio_coding/main/test/delay_test.cc b/modules/audio_coding/main/test/delay_test.cc
index 1a0f8f8..63bfe2b 100644
--- a/modules/audio_coding/main/test/delay_test.cc
+++ b/modules/audio_coding/main/test/delay_test.cc
@@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-
#include <assert.h>
#include <math.h>
@@ -17,8 +15,10 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
@@ -35,68 +35,76 @@
DEFINE_int32(delay, 0, "Delay in millisecond.");
DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
+DEFINE_bool(acm2, false, "Run the test with ACM2.");
+DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
+DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
namespace webrtc {
+
namespace {
-struct CodecConfig {
+struct CodecSettings {
char name[50];
int sample_rate_hz;
int num_channels;
};
-struct AcmConfig {
+struct AcmSettings {
bool dtx;
bool fec;
};
-struct Config {
- CodecConfig codec;
- AcmConfig acm;
+struct TestSettings {
+ CodecSettings codec;
+ AcmSettings acm;
bool packet_loss;
};
+} // namespace
class DelayTest {
public:
-
- DelayTest()
- : acm_a_(AudioCodingModule::Create(0)),
- acm_b_(AudioCodingModule::Create(1)),
- channel_a2b_(NULL),
+ explicit DelayTest(const Config& config)
+ : acm_a_(config.Get<AudioCodingModuleFactory>().Create(0)),
+ acm_b_(config.Get<AudioCodingModuleFactory>().Create(1)),
+ channel_a2b_(new Channel),
test_cntr_(0),
encoding_sample_rate_hz_(8000) {}
- ~DelayTest() {}
-
- void TearDown() {
+ ~DelayTest() {
if (channel_a2b_ != NULL) {
delete channel_a2b_;
channel_a2b_ = NULL;
}
+ in_file_a_.Close();
}
- void SetUp() {
+ void Initialize() {
test_cntr_ = 0;
std::string file_name = webrtc::test::ResourcePath(
"audio_coding/testfile32kHz", "pcm");
if (FLAGS_input_file.size() > 0)
file_name = FLAGS_input_file;
in_file_a_.Open(file_name, 32000, "rb");
- acm_a_->InitializeReceiver();
- acm_b_->InitializeReceiver();
+ ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
+ "Couldn't initialize receiver.\n";
+ ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
+ "Couldn't initialize receiver.\n";
if (FLAGS_init_delay > 0) {
- ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay));
+ ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
+ "Failed to set initial delay.\n";
}
if (FLAGS_delay > 0) {
- ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay));
+ ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
+ "Failed to set minimum delay.\n";
}
- uint8_t num_encoders = acm_a_->NumberOfCodecs();
+ int num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (int n = 0; n < num_encoders; n++) {
- acm_b_->Codec(n, &my_codec_param);
+ EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
+ "Failed to get codec.";
if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
my_codec_param.channels = 1;
else if (my_codec_param.channels > 1)
@@ -106,16 +114,17 @@
continue;
if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
continue;
- acm_b_->RegisterReceiveCodec(my_codec_param);
+ ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
+ "Couldn't register receive codec.\n";
}
// Create and connect the channel
- channel_a2b_ = new Channel;
- acm_a_->RegisterTransportCallback(channel_a2b_);
+ ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
+ "Couldn't register Transport callback.\n";
channel_a2b_->RegisterReceiverACM(acm_b_.get());
}
- void Perform(const Config* config, size_t num_tests, int duration_sec,
+ void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
const char* output_prefix) {
for (size_t n = 0; n < num_tests; ++n) {
ApplyConfig(config[n]);
@@ -124,8 +133,7 @@
}
private:
-
- void ApplyConfig(const Config& config) {
+ void ApplyConfig(const TestSettings& config) {
printf("====================================\n");
printf("Test %d \n"
"Codec: %s, %d kHz, %d channel(s)\n"
@@ -140,19 +148,22 @@
ConfigChannel(config.packet_loss);
}
- void SendCodec(const CodecConfig& config) {
+ void SendCodec(const CodecSettings& config) {
CodecInst my_codec_param;
- ASSERT_EQ(
- 0,
- AudioCodingModule::Codec(config.name, &my_codec_param,
- config.sample_rate_hz, config.num_channels));
+ ASSERT_EQ(0, AudioCodingModule::Codec(
+ config.name, &my_codec_param, config.sample_rate_hz,
+ config.num_channels)) << "Specified codec is not supported.\n";
+
encoding_sample_rate_hz_ = my_codec_param.plfreq;
- ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param));
+ ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
+ "Failed to register send-codec.\n";
}
- void ConfigAcm(const AcmConfig& config) {
- ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr));
- ASSERT_EQ(0, acm_a_->SetFECStatus(config.fec));
+ void ConfigAcm(const AcmSettings& config) {
+ ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
+ "Failed to set VAD.\n";
+ ASSERT_EQ(0, acm_a_->SetFECStatus(config.fec)) <<
+ "Failed to set FEC.\n";
}
void ConfigChannel(bool packet_loss) {
@@ -230,19 +241,39 @@
int encoding_sample_rate_hz_;
};
-void RunTest() {
- Config config;
- strcpy(config.codec.name, FLAGS_codec.c_str());
- config.codec.sample_rate_hz = FLAGS_sample_rate_hz;
- config.codec.num_channels = FLAGS_num_channels;
- config.acm.dtx = FLAGS_dtx;
- config.acm.fec = false;
- config.packet_loss = false;
-
- DelayTest delay_test;
- delay_test.SetUp();
- delay_test.Perform(&config, 1, 240, "delay_test");
- delay_test.TearDown();
-}
-} // namespace
} // namespace webrtc
+
+int main(int argc, char* argv[]) {
+ google::ParseCommandLineFlags(&argc, &argv, true);
+ webrtc::Config config;
+ webrtc::TestSettings test_setting;
+ strcpy(test_setting.codec.name, FLAGS_codec.c_str());
+
+ if (FLAGS_sample_rate_hz != 8000 &&
+ FLAGS_sample_rate_hz != 16000 &&
+ FLAGS_sample_rate_hz != 32000 &&
+ FLAGS_sample_rate_hz != 48000) {
+ std::cout << "Invalid sampling rate.\n";
+ return 1;
+ }
+ test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
+ if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
+ std::cout << "Only mono and stereo are supported.\n";
+ return 1;
+ }
+ test_setting.codec.num_channels = FLAGS_num_channels;
+ test_setting.acm.dtx = FLAGS_dtx;
+ test_setting.acm.fec = FLAGS_fec;
+ test_setting.packet_loss = FLAGS_packet_loss;
+
+ if (FLAGS_acm2) {
+ webrtc::UseNewAcm(&config);
+ } else {
+ webrtc::UseLegacyAcm(&config);
+ }
+
+ webrtc::DelayTest delay_test(config);
+ delay_test.Initialize();
+ delay_test.Perform(&test_setting, 1, 240, "delay_test");
+ return 0;
+}
