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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.

Browse Source 
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@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2013-11-18 18:16:53 +00:00
parent 9c5fb76662
commit 7a05ae5c69
2 changed files with 86 additions and 54 deletions
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5
webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
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@@ -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',
],

135
webrtc/modules/audio_coding/main/test/delay_test.cc
View File

@@ -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_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
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 @@ class DelayTest {
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 @@ class DelayTest {
}
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 @@ class DelayTest {
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 @@ class DelayTest {
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;
}