zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Added A/V sync tests with drifting clocks.

Browse Source 
adding 30% drift to media generator (e.g. audio frame generated every 7ms instead of promised 10ms) works fine
adding 2% drift to video ntp-timestamp-stamper makes A/V sync fail.

BUG=webrtc:5504
R=pbos@webrtc.org,stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1674413004

Cr-Commit-Position: refs/heads/master@{#11556}
This commit is contained in:
danilchap
2016-02-10 10:54:47 -08:00
committed by Commit bot
parent a75339c40d
commit 9c6a0c7f6d
8 changed files with 183 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

63
webrtc/call/call_perf_tests.cc
View File

@ -26,6 +26,7 @@
#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/drifting_clock.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
@ -41,11 +42,18 @@
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
using webrtc::test::DriftingClock;
using webrtc::test::FakeAudioDevice;
namespace webrtc {
class CallPerfTest : public test::CallTest {
protected:
void TestAudioVideoSync(bool fec, bool create_audio_first);
void TestAudioVideoSync(bool fec,
bool create_audio_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed);
void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
@ -188,7 +196,11 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
int64_t first_time_in_sync_;
};
void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
void CallPerfTest::TestAudioVideoSync(bool fec,
bool create_audio_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed) {
const char* kSyncGroup = "av_sync";
const uint32_t kAudioSendSsrc = 1234;
const uint32_t kAudioRecvSsrc = 5678;
@ -228,8 +240,8 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
const std::string audio_filename =
test::ResourcePath("voice_engine/audio_long16", "pcm");
ASSERT_STRNE("", audio_filename.c_str());
test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
audio_filename);
FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
audio_rtp_speed);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
Config voe_config;
voe_config.Set<VoicePacing>(new VoicePacing(true));
@ -324,7 +336,8 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
CreateFrameGeneratorCapturer();
DriftingClock drifting_clock(clock_, video_ntp_speed);
CreateFrameGeneratorCapturerWithDrift(&drifting_clock, video_rtp_speed);
Start();
@ -365,15 +378,49 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
TestAudioVideoSync(false, true);
TestAudioVideoSync(false, true, DriftingClock::kNoDrift,
DriftingClock::kNoDrift, DriftingClock::kNoDrift);
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
TestAudioVideoSync(false, false);
TestAudioVideoSync(false, false, DriftingClock::kNoDrift,
DriftingClock::kNoDrift, DriftingClock::kNoDrift);
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
TestAudioVideoSync(true, false);
TestAudioVideoSync(true, false, DriftingClock::kNoDrift,
DriftingClock::kNoDrift, DriftingClock::kNoDrift);
}
// TODO(danilchap): Reenable after adding support for frame capture clock
// that is not in sync with local TickTime clock.
TEST_F(CallPerfTest, DISABLED_PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
TestAudioVideoSync(false, true, DriftingClock::PercentsFaster(10.0f),
DriftingClock::kNoDrift, DriftingClock::kNoDrift);
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioRtpDrift) {
TestAudioVideoSync(false, true, DriftingClock::kNoDrift,
DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoRtpDrift) {
TestAudioVideoSync(false, true, DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f),
DriftingClock::kNoDrift);
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
TestAudioVideoSync(false, true, DriftingClock::kNoDrift,
DriftingClock::PercentsSlower(30.0f),
DriftingClock::PercentsFaster(30.0f));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
TestAudioVideoSync(false, true, DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f),
DriftingClock::PercentsSlower(30.0f));
}
void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,

14
webrtc/test/call_test.cc
View File

@ -236,6 +236,14 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
}
}
void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
float speed) {
VideoStream stream = video_encoder_config_.streams.back();
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
video_send_stream_->Input(), stream.width, stream.height,
stream.max_framerate * speed, clock));
}
void CallTest::CreateFrameGeneratorCapturer() {
VideoStream stream = video_encoder_config_.streams.back();
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
@ -245,9 +253,11 @@ void CallTest::CreateFrameGeneratorCapturer() {
void CallTest::CreateFakeAudioDevices() {
fake_send_audio_device_.reset(new FakeAudioDevice(
clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
DriftingClock::kNoDrift));
fake_recv_audio_device_.reset(new FakeAudioDevice(
clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
DriftingClock::kNoDrift));
}
void CallTest::CreateVideoStreams() {

1
webrtc/test/call_test.h
View File

@ -71,6 +71,7 @@ class CallTest : public ::testing::Test {
Transport* send_transport);
void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, float speed);
void CreateFrameGeneratorCapturer();
void CreateFakeAudioDevices();

57
webrtc/test/drifting_clock.cc Normal file
View File

@ -0,0 +1,57 @@
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/test/drifting_clock.h"
#include "webrtc/base/checks.h"
namespace webrtc {
namespace test {
const float DriftingClock::kDoubleSpeed = 2.0f;
const float DriftingClock::kNoDrift = 1.0f;
const float DriftingClock::kHalfSpeed = 0.5f;
DriftingClock::DriftingClock(Clock* clock, float speed)
: clock_(clock),
drift_(speed - 1.0f),
start_time_(clock_->TimeInMicroseconds()) {
RTC_CHECK(clock);
RTC_CHECK_GT(speed, 0.0f);
}
float DriftingClock::Drift() const {
int64_t now = clock_->TimeInMicroseconds();
RTC_DCHECK_GE(now, start_time_);
return (now - start_time_) * drift_;
}
int64_t DriftingClock::TimeInMilliseconds() const {
return clock_->TimeInMilliseconds() + Drift() / 1000.;
}
int64_t DriftingClock::TimeInMicroseconds() const {
return clock_->TimeInMicroseconds() + Drift();
}
void DriftingClock::CurrentNtp(uint32_t& seconds, uint32_t& fractions) const {
// NTP precision is 1/2^32 seconds, i.e. 2^32 ntp fractions = 1 second.
const double kNtpFracPerMicroSecond = 4294.967296; // = 2^32 / 10^6
clock_->CurrentNtp(seconds, fractions);
uint64_t total_fractions = (static_cast<uint64_t>(seconds) << 32) | fractions;
total_fractions += Drift() * kNtpFracPerMicroSecond;
seconds = total_fractions >> 32;
fractions = static_cast<uint32_t>(total_fractions);
}
int64_t DriftingClock::CurrentNtpInMilliseconds() const {
return clock_->CurrentNtpInMilliseconds() + Drift() / 1000.;
}
} // namespace test
} // namespace webrtc

45
webrtc/test/drifting_clock.h Normal file
View File

@ -0,0 +1,45 @@
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_TEST_DRIFTING_CLOCK_H_
#define WEBRTC_TEST_DRIFTING_CLOCK_H_
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
namespace test {
class DriftingClock : public Clock {
public:
// TODO(danilchap): Make this constants constexpr when it would be supported.
static const float kDoubleSpeed; // 2.0f;
static const float kNoDrift; // 1.0f;
static const float kHalfSpeed; // 0.5f;
DriftingClock(Clock* clock, float speed);
// TODO(danilchap): Make this functions constexpr when it would be supported.
static float PercentsFaster(float percent) { return 1.0f + percent / 100.0f; }
static float PercentsSlower(float percent) { return 1.0f - percent / 100.0f; }
int64_t TimeInMilliseconds() const override;
int64_t TimeInMicroseconds() const override;
void CurrentNtp(uint32_t& seconds, uint32_t& fractions) const override;
int64_t CurrentNtpInMilliseconds() const override;
private:
float Drift() const;
Clock* const clock_;
const float drift_;
const int64_t start_time_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_DRIFTING_CLOCK_H_

11
webrtc/test/fake_audio_device.cc
View File

@ -22,13 +22,16 @@
namespace webrtc {
namespace test {
FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
FakeAudioDevice::FakeAudioDevice(Clock* clock,
const std::string& filename,
float speed)
: audio_callback_(NULL),
capturing_(false),
captured_audio_(),
playout_buffer_(),
speed_(speed),
last_playout_ms_(-1),
clock_(clock),
clock_(clock, speed),
tick_(EventTimerWrapper::Create()),
thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
file_utility_(new ModuleFileUtility(0)),
@ -51,7 +54,7 @@ int32_t FakeAudioDevice::Init() {
if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
return -1;
if (!tick_->StartTimer(true, 10))
if (!tick_->StartTimer(true, 10 / speed_))
return -1;
thread_.Start();
thread_.SetPriority(rtc::kHighPriority);
@ -107,7 +110,7 @@ void FakeAudioDevice::CaptureAudio() {
false,
new_mic_level));
size_t samples_needed = kFrequencyHz / 100;
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t now_ms = clock_.TimeInMilliseconds();
uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
samples_needed = std::min(

6
webrtc/test/fake_audio_device.h
View File

@ -16,6 +16,7 @@
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
#include "webrtc/test/drifting_clock.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -29,7 +30,7 @@ namespace test {
class FakeAudioDevice : public FakeAudioDeviceModule {
public:
FakeAudioDevice(Clock* clock, const std::string& filename);
FakeAudioDevice(Clock* clock, const std::string& filename, float speed);
virtual ~FakeAudioDevice();
@ -54,9 +55,10 @@ class FakeAudioDevice : public FakeAudioDeviceModule {
bool capturing_;
int8_t captured_audio_[kBufferSizeBytes];
int8_t playout_buffer_[kBufferSizeBytes];
const float speed_;
int64_t last_playout_ms_;
Clock* clock_;
DriftingClock clock_;
rtc::scoped_ptr<EventTimerWrapper> tick_;
rtc::CriticalSection lock_;
rtc::PlatformThread thread_;

2
webrtc/test/webrtc_test_common.gyp
View File

@ -22,6 +22,8 @@
'constants.h',
'direct_transport.cc',
'direct_transport.h',
'drifting_clock.cc',
'drifting_clock.h',
'encoder_settings.cc',
'encoder_settings.h',
'fake_audio_device.cc',