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Let A/V sync test use default AudioCoding module

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This test used to run with both ACM1 and ACM2, to verify sync with both
versions of the module. ACM1 (and NetEq3) is now being deprecated,
wherefore this test should now use the default one (i.e., ACM2).

BUG=2996
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5953 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-04-22 08:36:33 +00:00
parent 0c1444c748
commit d144bb6812
1 changed files with 7 additions and 36 deletions
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43
webrtc/video/call_perf_tests.cc
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@ -16,7 +16,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/call.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
@ -156,16 +155,14 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
VideoRtcpAndSyncObserver(Clock* clock,
int voe_channel,
VoEVideoSync* voe_sync,
SyncRtcpObserver* audio_observer,
bool using_new_acm)
SyncRtcpObserver* audio_observer)
: SyncRtcpObserver(FakeNetworkPipe::Config()),
clock_(clock),
voe_channel_(voe_channel),
voe_sync_(voe_sync),
audio_observer_(audio_observer),
creation_time_ms_(clock_->TimeInMilliseconds()),
first_time_in_sync_(-1),
using_new_acm_(using_new_acm) {}
first_time_in_sync_(-1) {}
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
@ -184,12 +181,8 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
int64_t stream_offset = latest_audio_ntp - latest_video_ntp;
std::stringstream ss;
ss << stream_offset;
std::stringstream acm_type;
if (using_new_acm_) {
acm_type << "_acm2";
}
webrtc::test::PrintResult("stream_offset",
acm_type.str(),
"",
"synchronization",
ss.str(),
"ms",
@ -203,7 +196,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time",
acm_type.str(),
"",
"synchronization",
time_since_creation,
"ms",
@ -221,19 +214,9 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
SyncRtcpObserver* audio_observer_;
int64_t creation_time_ms_;
int64_t first_time_in_sync_;
bool using_new_acm_;
};
class ParamCallPerfTest : public CallPerfTest,
public ::testing::WithParamInterface<bool> {
public:
ParamCallPerfTest() : CallPerfTest(), use_new_acm_(GetParam()) {}
protected:
bool use_new_acm_;
};
TEST_P(ParamCallPerfTest, PlaysOutAudioAndVideoInSync) {
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
VoiceEngine* voice_engine = VoiceEngine::Create();
VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
@ -245,15 +228,7 @@ TEST_P(ParamCallPerfTest, PlaysOutAudioAndVideoInSync) {
test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
audio_filename);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL));
Config config;
if (use_new_acm_) {
config.Set<webrtc::AudioCodingModuleFactory>(
new webrtc::NewAudioCodingModuleFactory());
} else {
config.Set<webrtc::AudioCodingModuleFactory>(
new webrtc::AudioCodingModuleFactory());
}
int channel = voe_base->CreateChannel(config);
int channel = voe_base->CreateChannel();
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 500;
@ -261,8 +236,7 @@ TEST_P(ParamCallPerfTest, PlaysOutAudioAndVideoInSync) {
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
channel,
voe_sync,
&audio_observer,
use_new_acm_);
&audio_observer);
Call::Config receiver_config(observer.ReceiveTransport());
receiver_config.voice_engine = voice_engine;
@ -369,9 +343,6 @@ TEST_P(ParamCallPerfTest, PlaysOutAudioAndVideoInSync) {
VoiceEngine::Delete(voice_engine);
}
// Test with both ACM1 and ACM2.
INSTANTIATE_TEST_CASE_P(SwitchAcm, ParamCallPerfTest, ::testing::Bool());
TEST_F(CallPerfTest, RegisterCpuOveruseObserver) {
// Verifies that either a normal or overuse callback is triggered.
class OveruseCallbackObserver : public test::RtpRtcpObserver,