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NetEq4: Changing the behavior of playout_timestamp_ update

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The variable playout_timestamp_ was not updated to the latest decoded
timestamp while comfort noise was played. Instead, it was upadted using
dead reckoning, which caused it to drift away from the timestamps of the
incoming CNG packets. Now it is updated also during comfort noise
playout.

Since the change is only in NetEq4, this change also makes the test
PlaysOutAudioAndVideoInSync use both ACM1/NetEq3 and ACM2/NetEq4.

Re-enabling one NetEq unit test that is no longer failing thanks to this CL.

BUG=2932
R=stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5649 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-03-06 10:28:07 +00:00
parent 60ad5fdadf
commit ed865b5d46
3 changed files with 51 additions and 16 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_coding/neteq4/neteq_impl.cc
View File

@ -833,10 +833,10 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
sync_buffer_->set_dtmf_index(sync_buffer_->Size());
}
if ((last_mode_ != kModeExpand) && (last_mode_ != kModeRfc3389Cng)) {
// If last operation was neither expand, nor comfort noise, calculate the
// |playout_timestamp_| from the |sync_buffer_|. However, do not update the
// |playout_timestamp_| if it would be moved "backwards".
if (last_mode_ != kModeExpand) {
// If last operation was not expand, calculate the |playout_timestamp_| from
// the |sync_buffer_|. However, do not update the |playout_timestamp_| if it
// would be moved "backwards".
uint32_t temp_timestamp = sync_buffer_->end_timestamp() -
static_cast<uint32_t>(sync_buffer_->FutureLength());
if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) {

5
webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
View File

@ -788,15 +788,14 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor) {
EXPECT_GE(delay_after, delay_before - 20 * 16);
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithClockNegativeDrift)) {
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithNegativeClockDrift)) {
// Apply a clock drift of -25 ms / s (sender faster than receiver).
const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
LongCngWithClockDrift(kDriftFactor);
}
// TODO(hlundin): Re-enable this test and fix the issues to make it pass.
TEST_F(NetEqDecodingTest,
DISABLED_ON_ANDROID(DISABLED_LongCngWithClockPositiveDrift)) {
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithPositiveClockDrift)) {
// Apply a clock drift of +25 ms / s (sender slower than receiver).
const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
LongCngWithClockDrift(kDriftFactor);

54
webrtc/video/call_perf_tests.cc
View File

@ -16,6 +16,8 @@
#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"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@ -151,14 +153,16 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
VideoRtcpAndSyncObserver(Clock* clock,
int voe_channel,
VoEVideoSync* voe_sync,
SyncRtcpObserver* audio_observer)
SyncRtcpObserver* audio_observer,
bool using_new_acm)
: 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) {}
first_time_in_sync_(-1),
using_new_acm_(using_new_acm) {}
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
@ -177,8 +181,16 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
int64_t stream_offset = latest_audio_ntp - latest_video_ntp;
std::stringstream ss;
ss << stream_offset;
webrtc::test::PrintResult(
"stream_offset", "", "synchronization", ss.str(), "ms", false);
std::stringstream acm_type;
if (using_new_acm_) {
acm_type << "_acm2";
}
webrtc::test::PrintResult("stream_offset",
acm_type.str(),
"synchronization",
ss.str(),
"ms",
false);
int64_t time_since_creation = now_ms - creation_time_ms_;
// During the first couple of seconds audio and video can falsely be
// estimated as being synchronized. We don't want to trigger on those.
@ -188,7 +200,7 @@ 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",
@ -206,9 +218,19 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
SyncRtcpObserver* audio_observer_;
int64_t creation_time_ms_;
int64_t first_time_in_sync_;
bool using_new_acm_;
};
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
class ParamCallPerfTest : public CallPerfTest,
public ::testing::WithParamInterface<bool> {
public:
ParamCallPerfTest() : CallPerfTest(), use_new_acm_(GetParam()) {}
protected:
bool use_new_acm_;
};
TEST_P(ParamCallPerfTest, PlaysOutAudioAndVideoInSync) {
VoiceEngine* voice_engine = VoiceEngine::Create();
VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
@ -220,13 +242,24 @@ TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
audio_filename);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL));
int channel = voe_base->CreateChannel();
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);
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 500;
SyncRtcpObserver audio_observer(net_config);
VideoRtcpAndSyncObserver observer(
Clock::GetRealTimeClock(), channel, voe_sync, &audio_observer);
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
channel,
voe_sync,
&audio_observer,
use_new_acm_);
Call::Config receiver_config(observer.ReceiveTransport());
receiver_config.voice_engine = voice_engine;
@ -329,6 +362,9 @@ TEST_F(CallPerfTest, 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,