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Removing AudioCoding duplicate tests

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Reverting to using one version of ACM in ACM tests.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-04-17 08:29:10 +00:00
parent 6cec07f6a7
commit adaf809612
23 changed files with 197 additions and 548 deletions
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16
webrtc/modules/audio_coding/main/test/delay_test.cc
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@ -35,7 +35,6 @@ 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).");
@ -64,9 +63,9 @@ struct TestSettings {
class DelayTest {
public:
explicit DelayTest(const Config& config)
: acm_a_(config.Get<AudioCodingModuleFactory>().Create(0)),
acm_b_(config.Get<AudioCodingModuleFactory>().Create(1)),
DelayTest()
: acm_a_(AudioCodingModule::Create(0)),
acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(new Channel),
test_cntr_(0),
encoding_sample_rate_hz_(8000) {}
@ -245,7 +244,6 @@ class DelayTest {
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());
@ -266,13 +264,7 @@ int main(int argc, char* argv[]) {
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);
webrtc::DelayTest delay_test;
delay_test.Initialize();
delay_test.Perform(&test_setting, 1, 240, "delay_test");
return 0;