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Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

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This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2014-07-11 13:44:02 +00:00
parent ccbed3b3c4
commit 168f23faa5
24 changed files with 168 additions and 134 deletions
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1
webrtc/build/tsan_suppressions.cc
View File

@ -71,7 +71,6 @@ char kTSanDefaultSuppressions[] =
"deadlock:webrtc::RTPSenderAudio::RegisterAudioPayload\n"
"deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n"
"deadlock:webrtc::vcm::VideoReceiver::RegisterPacketRequestCallback\n"
"deadlock:webrtc::VideoSendStreamTest_SuspendBelowMinBitrate_Test::TestBody\n"
"deadlock:webrtc::ViECaptureImpl::ConnectCaptureDevice\n"
"deadlock:webrtc::ViEChannel::StartSend\n"
"deadlock:webrtc::ViECodecImpl::GetSendSideDelay\n"

9
webrtc/common_types.h
View File

@ -269,6 +269,15 @@ class FrameCountObserver {
const unsigned int ssrc) = 0;
};
// Callback, used to notify an observer whenever the send-side delay is updated.
class SendSideDelayObserver {
public:
virtual ~SendSideDelayObserver() {}
virtual void SendSideDelayUpdated(int avg_delay_ms,
int max_delay_ms,
uint32_t ssrc) = 0;
};
// ==================================================================
// Voice specific types
// ==================================================================

9
webrtc/config.h
View File

@ -34,10 +34,17 @@ struct RtpStatistics {
};
struct StreamStats {
StreamStats() : key_frames(0), delta_frames(0), bitrate_bps(0) {}
StreamStats()
: key_frames(0),
delta_frames(0),
bitrate_bps(0),
avg_delay_ms(0),
max_delay_ms(0) {}
uint32_t key_frames;
uint32_t delta_frames;
int32_t bitrate_bps;
int avg_delay_ms;
int max_delay_ms;
StreamDataCounters rtp_stats;
RtcpStatistics rtcp_stats;
};

5
webrtc/modules/pacing/include/paced_sender.h
View File

@ -17,7 +17,6 @@
#include "webrtc/modules/interface/module.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -147,8 +146,8 @@ class PacedSender : public Module {
scoped_ptr<paced_sender::IntervalBudget> padding_budget_
GUARDED_BY(critsect_);
TickTime time_last_update_ GUARDED_BY(critsect_);
TickTime time_last_send_ GUARDED_BY(critsect_);
int64_t time_last_update_ GUARDED_BY(critsect_);
int64_t time_last_send_ GUARDED_BY(critsect_);
int64_t capture_time_ms_last_queued_ GUARDED_BY(critsect_);
int64_t capture_time_ms_last_sent_ GUARDED_BY(critsect_);

13
webrtc/modules/pacing/paced_sender.cc
View File

@ -142,7 +142,7 @@ PacedSender::PacedSender(Clock* clock,
max_queue_length_ms_(kDefaultMaxQueueLengthMs),
media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)),
padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)),
time_last_update_(TickTime::Now()),
time_last_update_(clock->TimeInMilliseconds()),
capture_time_ms_last_queued_(0),
capture_time_ms_last_sent_(0),
high_priority_packets_(new paced_sender::PacketList),
@ -248,8 +248,7 @@ int PacedSender::QueueInMs() const {
int32_t PacedSender::TimeUntilNextProcess() {
CriticalSectionScoped cs(critsect_.get());
int64_t elapsed_time_ms =
(TickTime::Now() - time_last_update_).Milliseconds();
int64_t elapsed_time_ms = clock_->TimeInMilliseconds() - time_last_update_;
if (elapsed_time_ms <= 0) {
return kMinPacketLimitMs;
}
@ -260,9 +259,9 @@ int32_t PacedSender::TimeUntilNextProcess() {
}
int32_t PacedSender::Process() {
TickTime now = TickTime::Now();
int64_t now = clock_->TimeInMilliseconds();
CriticalSectionScoped cs(critsect_.get());
int elapsed_time_ms = (now - time_last_update_).Milliseconds();
int elapsed_time_ms = now - time_last_update_;
time_last_update_ = now;
if (!enabled_) {
return 0;
@ -335,7 +334,7 @@ bool PacedSender::ShouldSendNextPacket(paced_sender::PacketList** packet_list) {
if (media_budget_->bytes_remaining() <= 0) {
// All bytes consumed for this interval.
// Check if we have not sent in a too long time.
if ((TickTime::Now() - time_last_send_).Milliseconds() >
if (clock_->TimeInMilliseconds() - time_last_send_ >
kMaxQueueTimeWithoutSendingMs) {
if (!high_priority_packets_->empty()) {
*packet_list = high_priority_packets_.get();
@ -389,7 +388,7 @@ paced_sender::Packet PacedSender::GetNextPacketFromList(
// MUST have critsect_ when calling.
void PacedSender::UpdateMediaBytesSent(int num_bytes) {
time_last_send_ = TickTime::Now();
time_last_send_ = clock_->TimeInMilliseconds();
media_budget_->UseBudget(num_bytes);
padding_budget_->UseBudget(num_bytes);
}

29
webrtc/modules/pacing/paced_sender_unittest.cc
View File

@ -58,7 +58,6 @@ class PacedSenderTest : public ::testing::Test {
protected:
PacedSenderTest() : clock_(123456) {
srand(0);
TickTime::UseFakeClock(123456);
// Need to initialize PacedSender after we initialize clock.
send_bucket_.reset(
new PacedSender(
@ -99,10 +98,8 @@ TEST_F(PacedSenderTest, QueuePacket) {
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
clock_.AdvanceTimeMilliseconds(4);
TickTime::AdvanceFakeClock(4);
EXPECT_EQ(1, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(1);
TickTime::AdvanceFakeClock(1);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_CALL(callback_, TimeToSendPacket(
ssrc, sequence_number++, queued_packet_timestamp, false))
@ -137,7 +134,6 @@ TEST_F(PacedSenderTest, PaceQueuedPackets) {
for (int k = 0; k < 10; ++k) {
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_CALL(callback_,
TimeToSendPacket(ssrc, _, _, false))
.Times(3)
@ -147,7 +143,6 @@ TEST_F(PacedSenderTest, PaceQueuedPackets) {
}
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
@ -185,7 +180,6 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
for (int k = 0; k < 10; ++k) {
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
for (int i = 0; i < 3; ++i) {
EXPECT_CALL(callback_, TimeToSendPacket(ssrc, queued_sequence_number++,
@ -199,7 +193,6 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
}
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
@ -233,7 +226,6 @@ TEST_F(PacedSenderTest, CanQueuePacketsWithSameSequenceNumberOnDifferentSsrcs) {
false);
clock_.AdvanceTimeMilliseconds(1000);
TickTime::AdvanceFakeClock(1000);
send_bucket_->Process();
}
@ -253,7 +245,6 @@ TEST_F(PacedSenderTest, Padding) {
EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
@ -262,7 +253,6 @@ TEST_F(PacedSenderTest, Padding) {
WillOnce(Return(250));
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
}
@ -274,13 +264,11 @@ TEST_F(PacedSenderTest, NoPaddingWhenDisabled) {
EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
}
@ -297,7 +285,6 @@ TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) {
SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
capture_time_ms, 250, false);
clock_.AdvanceTimeMilliseconds(kTimeStep);
TickTime::AdvanceFakeClock(kTimeStep);
EXPECT_CALL(callback_, TimeToSendPadding(250)).Times(1).
WillOnce(Return(250));
send_bucket_->Process();
@ -323,7 +310,6 @@ TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
media_payload, false));
media_bytes += media_payload;
clock_.AdvanceTimeMilliseconds(kTimeStep);
TickTime::AdvanceFakeClock(kTimeStep);
send_bucket_->Process();
}
EXPECT_NEAR(kTargetBitrate, 8 * (media_bytes + callback.padding_sent()) /
@ -365,7 +351,6 @@ TEST_F(PacedSenderTest, Priority) {
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
@ -376,7 +361,6 @@ TEST_F(PacedSenderTest, Priority) {
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
}
@ -408,7 +392,6 @@ TEST_F(PacedSenderTest, Pause) {
ssrc, sequence_number++, capture_time_ms, 250, false));
clock_.AdvanceTimeMilliseconds(10000);
TickTime::AdvanceFakeClock(10000);
int64_t second_capture_time_ms = clock_.TimeInMilliseconds();
// Expect everything to be queued.
@ -425,7 +408,6 @@ TEST_F(PacedSenderTest, Pause) {
for (int i = 0; i < 10; ++i) {
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
}
@ -438,7 +420,6 @@ TEST_F(PacedSenderTest, Pause) {
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
@ -448,7 +429,6 @@ TEST_F(PacedSenderTest, Pause) {
.WillRepeatedly(Return(true));
EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
clock_.AdvanceTimeMilliseconds(5);
TickTime::AdvanceFakeClock(5);
EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
EXPECT_EQ(0, send_bucket_->Process());
EXPECT_EQ(0, send_bucket_->QueueInMs());
@ -467,7 +447,6 @@ TEST_F(PacedSenderTest, ResendPacket) {
250,
false));
clock_.AdvanceTimeMilliseconds(1);
TickTime::AdvanceFakeClock(1);
EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
ssrc,
sequence_number + 1,
@ -475,7 +454,6 @@ TEST_F(PacedSenderTest, ResendPacket) {
250,
false));
clock_.AdvanceTimeMilliseconds(9999);
TickTime::AdvanceFakeClock(9999);
EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms,
send_bucket_->QueueInMs());
// Fails to send first packet so only one call.
@ -484,7 +462,6 @@ TEST_F(PacedSenderTest, ResendPacket) {
.Times(1)
.WillOnce(Return(false));
clock_.AdvanceTimeMilliseconds(10000);
TickTime::AdvanceFakeClock(10000);
send_bucket_->Process();
// Queue remains unchanged.
@ -501,7 +478,6 @@ TEST_F(PacedSenderTest, ResendPacket) {
.Times(1)
.WillOnce(Return(false));
clock_.AdvanceTimeMilliseconds(10000);
TickTime::AdvanceFakeClock(10000);
send_bucket_->Process();
// Queue is reduced by 1 packet.
@ -514,7 +490,6 @@ TEST_F(PacedSenderTest, ResendPacket) {
.Times(1)
.WillOnce(Return(true));
clock_.AdvanceTimeMilliseconds(10000);
TickTime::AdvanceFakeClock(10000);
send_bucket_->Process();
EXPECT_EQ(0, send_bucket_->QueueInMs());
}
@ -535,7 +510,6 @@ TEST_F(PacedSenderTest, MaxQueueLength) {
}
clock_.AdvanceTimeMilliseconds(2001);
TickTime::AdvanceFakeClock(2001);
SendAndExpectPacket(PacedSender::kNormalPriority,
ssrc,
sequence_number++,
@ -546,7 +520,7 @@ TEST_F(PacedSenderTest, MaxQueueLength) {
send_bucket_->Process();
EXPECT_EQ(0, send_bucket_->QueueInMs());
clock_.AdvanceTimeMilliseconds(31);
TickTime::AdvanceFakeClock(31);
send_bucket_->Process();
}
@ -564,7 +538,6 @@ TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) {
false);
clock_.AdvanceTimeMilliseconds(500);
TickTime::AdvanceFakeClock(500);
EXPECT_EQ(500, send_bucket_->QueueInMs());
send_bucket_->Process();
EXPECT_EQ(0, send_bucket_->QueueInMs());

5
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
View File

@ -47,8 +47,8 @@ class RtpRtcp : public Module {
* intra_frame_callback - Called when the receiver request a intra frame.
* bandwidth_callback - Called when we receive a changed estimate from
* the receiver of out stream.
* audio_messages - Telehone events. May not be NULL; default callback
* will do nothing.
* audio_messages - Telephone events. May not be NULL; default
* callback will do nothing.
* remote_bitrate_estimator - Estimates the bandwidth available for a set of
* streams from the same client.
* paced_sender - Spread any bursts of packets into smaller
@ -69,6 +69,7 @@ class RtpRtcp : public Module {
PacedSender* paced_sender;
BitrateStatisticsObserver* send_bitrate_observer;
FrameCountObserver* send_frame_count_observer;
SendSideDelayObserver* send_side_delay_observer;
};
/*

6
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -39,7 +39,8 @@ RtpRtcp::Configuration::Configuration()
remote_bitrate_estimator(NULL),
paced_sender(NULL),
send_bitrate_observer(NULL),
send_frame_count_observer(NULL) {
send_frame_count_observer(NULL),
send_side_delay_observer(NULL) {
}
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
@ -64,7 +65,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.audio_messages,
configuration.paced_sender,
configuration.send_bitrate_observer,
configuration.send_frame_count_observer),
configuration.send_frame_count_observer,
configuration.send_side_delay_observer),
rtcp_sender_(configuration.id,
configuration.audio,
configuration.clock,

32
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -47,7 +47,8 @@ RTPSender::RTPSender(const int32_t id,
RtpAudioFeedback* audio_feedback,
PacedSender* paced_sender,
BitrateStatisticsObserver* bitrate_callback,
FrameCountObserver* frame_count_observer)
FrameCountObserver* frame_count_observer,
SendSideDelayObserver* send_side_delay_observer)
: clock_(clock),
bitrate_sent_(clock, this),
id_(id),
@ -75,6 +76,7 @@ RTPSender::RTPSender(const int32_t id,
rtp_stats_callback_(NULL),
bitrate_callback_(bitrate_callback),
frame_count_observer_(frame_count_observer),
send_side_delay_observer_(send_side_delay_observer),
// RTP variables
start_timestamp_forced_(false),
start_timestamp_(0),
@ -164,9 +166,7 @@ uint32_t RTPSender::NackOverheadRate() const {
bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms,
int* max_send_delay_ms) const {
if (!SendingMedia())
return false;
CriticalSectionScoped cs(statistics_crit_.get());
CriticalSectionScoped lock(statistics_crit_.get());
SendDelayMap::const_iterator it = send_delays_.upper_bound(
clock_->TimeInMilliseconds() - kSendSideDelayWindowMs);
if (it == send_delays_.end())
@ -997,10 +997,26 @@ int32_t RTPSender::SendToNetwork(
}
void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
CriticalSectionScoped cs(statistics_crit_.get());
send_delays_[now_ms] = now_ms - capture_time_ms;
send_delays_.erase(send_delays_.begin(),
send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs));
uint32_t ssrc;
int avg_delay_ms = 0;
int max_delay_ms = 0;
{
CriticalSectionScoped lock(send_critsect_);
ssrc = ssrc_;
}
{
CriticalSectionScoped cs(statistics_crit_.get());
// TODO(holmer): Compute this iteratively instead.
send_delays_[now_ms] = now_ms - capture_time_ms;
send_delays_.erase(send_delays_.begin(),
send_delays_.lower_bound(now_ms -
kSendSideDelayWindowMs));
}
if (send_side_delay_observer_ &&
GetSendSideDelay(&avg_delay_ms, &max_delay_ms)) {
send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms,
max_delay_ms, ssrc);
}
}
void RTPSender::ProcessBitrate() {

4
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -71,7 +71,8 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
Transport *transport, RtpAudioFeedback *audio_feedback,
PacedSender *paced_sender,
BitrateStatisticsObserver* bitrate_callback,
FrameCountObserver* frame_count_observer);
FrameCountObserver* frame_count_observer,
SendSideDelayObserver* send_side_delay_observer);
virtual ~RTPSender();
void ProcessBitrate();
@ -379,6 +380,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
BitrateStatisticsObserver* const bitrate_callback_;
FrameCountObserver* const frame_count_observer_;
SendSideDelayObserver* const send_side_delay_observer_;
// RTP variables
bool start_timestamp_forced_ GUARDED_BY(send_critsect_);

10
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -94,7 +94,7 @@ class RtpSenderTest : public ::testing::Test {
virtual void SetUp() {
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, NULL, NULL));
&mock_paced_sender_, NULL, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
@ -672,7 +672,7 @@ TEST_F(RtpSenderTest, SendPadding) {
TEST_F(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
&mock_paced_sender_, NULL, NULL));
&mock_paced_sender_, NULL, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
// Make all packets go through the pacer.
EXPECT_CALL(mock_paced_sender_,
@ -818,7 +818,7 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
} callback;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, NULL, &callback));
&mock_paced_sender_, NULL, &callback, NULL));
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
@ -867,7 +867,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
BitrateStatistics bitrate_;
} callback;
rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, &callback, NULL));
&mock_paced_sender_, &callback, NULL, NULL));
// Simulate kNumPackets sent with kPacketInterval ms intervals.
const uint32_t kNumPackets = 15;
@ -923,7 +923,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
virtual void SetUp() {
payload_ = kAudioPayload;
rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL,
&mock_paced_sender_, NULL, NULL));
&mock_paced_sender_, NULL, NULL, NULL));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};

6
webrtc/video/end_to_end_tests.cc
View File

@ -1345,9 +1345,6 @@ TEST_F(EndToEndTest, GetStats) {
send_stats_filled_["NumStreams"] |=
stats.substreams.size() == expected_send_ssrcs_.size();
send_stats_filled_["Delay"] |=
stats.avg_delay_ms != 0 || stats.max_delay_ms != 0;
for (std::map<uint32_t, StreamStats>::const_iterator it =
stats.substreams.begin();
it != stats.substreams.end();
@ -1380,6 +1377,9 @@ TEST_F(EndToEndTest, GetStats) {
send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
stats.encode_frame_rate != 0;
send_stats_filled_[CompoundKey("Delay", it->first)] |=
stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
}
return AllStatsFilled(send_stats_filled_);

24
webrtc/video/send_statistics_proxy.cc
View File

@ -17,10 +17,8 @@
namespace webrtc {
SendStatisticsProxy::SendStatisticsProxy(
const VideoSendStream::Config& config,
SendStatisticsProxy::StatsProvider* stats_provider)
const VideoSendStream::Config& config)
: config_(config),
stats_provider_(stats_provider),
crit_(CriticalSectionWrapper::CreateCriticalSection()) {
}
@ -45,13 +43,8 @@ void SendStatisticsProxy::CapturedFrameRate(const int capture_id,
}
VideoSendStream::Stats SendStatisticsProxy::GetStats() const {
VideoSendStream::Stats stats;
{
CriticalSectionScoped lock(crit_.get());
stats = stats_;
}
stats_provider_->GetSendSideDelay(&stats);
return stats;
CriticalSectionScoped lock(crit_.get());
return stats_;
}
StreamStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
@ -119,4 +112,15 @@ void SendStatisticsProxy::FrameCountUpdated(FrameType frame_type,
}
}
void SendStatisticsProxy::SendSideDelayUpdated(int avg_delay_ms,
int max_delay_ms,
uint32_t ssrc) {
CriticalSectionScoped lock(crit_.get());
StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
stats->avg_delay_ms = avg_delay_ms;
stats->max_delay_ms = max_delay_ms;
}
} // namespace webrtc

20
webrtc/video/send_statistics_proxy.h
View File

@ -29,19 +29,10 @@ class SendStatisticsProxy : public RtcpStatisticsCallback,
public BitrateStatisticsObserver,
public FrameCountObserver,
public ViEEncoderObserver,
public ViECaptureObserver {
public ViECaptureObserver,
public SendSideDelayObserver {
public:
class StatsProvider {
protected:
StatsProvider() {}
virtual ~StatsProvider() {}
public:
virtual bool GetSendSideDelay(VideoSendStream::Stats* stats) = 0;
};
SendStatisticsProxy(const VideoSendStream::Config& config,
StatsProvider* stats_provider);
explicit SendStatisticsProxy(const VideoSendStream::Config& config);
virtual ~SendStatisticsProxy();
VideoSendStream::Stats GetStats() const;
@ -79,11 +70,14 @@ class SendStatisticsProxy : public RtcpStatisticsCallback,
virtual void NoPictureAlarm(const int capture_id,
const CaptureAlarm alarm) OVERRIDE {}
virtual void SendSideDelayUpdated(int avg_delay_ms,
int max_delay_ms,
uint32_t ssrc) OVERRIDE;
private:
StreamStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
const VideoSendStream::Config config_;
StatsProvider* const stats_provider_;
scoped_ptr<CriticalSectionWrapper> crit_;
VideoSendStream::Stats stats_ GUARDED_BY(crit_);
};

38
webrtc/video/send_statistics_proxy_unittest.cc
View File

@ -19,8 +19,7 @@
namespace webrtc {
class SendStatisticsProxyTest : public ::testing::Test,
protected SendStatisticsProxy::StatsProvider {
class SendStatisticsProxyTest : public ::testing::Test {
public:
SendStatisticsProxyTest() : avg_delay_ms_(0), max_delay_ms_(0) {}
virtual ~SendStatisticsProxyTest() {}
@ -28,7 +27,7 @@ class SendStatisticsProxyTest : public ::testing::Test,
protected:
virtual void SetUp() {
statistics_proxy_.reset(
new SendStatisticsProxy(GetTestConfig(), this));
new SendStatisticsProxy(GetTestConfig()));
config_ = GetTestConfig();
expected_ = VideoSendStream::Stats();
}
@ -40,18 +39,9 @@ class SendStatisticsProxyTest : public ::testing::Test,
return config;
}
virtual bool GetSendSideDelay(VideoSendStream::Stats* stats) OVERRIDE {
stats->avg_delay_ms = avg_delay_ms_;
stats->max_delay_ms = max_delay_ms_;
return true;
}
void ExpectEqual(VideoSendStream::Stats one, VideoSendStream::Stats other) {
EXPECT_EQ(one.avg_delay_ms, other.avg_delay_ms);
EXPECT_EQ(one.input_frame_rate, other.input_frame_rate);
EXPECT_EQ(one.encode_frame_rate, other.encode_frame_rate);
EXPECT_EQ(one.avg_delay_ms, other.avg_delay_ms);
EXPECT_EQ(one.max_delay_ms, other.max_delay_ms);
EXPECT_EQ(one.suspended, other.suspended);
EXPECT_EQ(one.substreams.size(), other.substreams.size());
@ -68,6 +58,8 @@ class SendStatisticsProxyTest : public ::testing::Test,
EXPECT_EQ(a.key_frames, b.key_frames);
EXPECT_EQ(a.delta_frames, b.delta_frames);
EXPECT_EQ(a.bitrate_bps, b.bitrate_bps);
EXPECT_EQ(a.avg_delay_ms, b.avg_delay_ms);
EXPECT_EQ(a.max_delay_ms, b.max_delay_ms);
EXPECT_EQ(a.rtp_stats.bytes, b.rtp_stats.bytes);
EXPECT_EQ(a.rtp_stats.header_bytes, b.rtp_stats.header_bytes);
@ -190,6 +182,7 @@ TEST_F(SendStatisticsProxyTest, Bitrate) {
++it) {
const uint32_t ssrc = *it;
BitrateStatistics bitrate;
// Use ssrc as bitrate_bps to get a unique value for each stream.
bitrate.bitrate_bps = ssrc;
observer->Notify(bitrate, ssrc);
expected_.substreams[ssrc].bitrate_bps = ssrc;
@ -199,14 +192,23 @@ TEST_F(SendStatisticsProxyTest, Bitrate) {
ExpectEqual(expected_, stats);
}
TEST_F(SendStatisticsProxyTest, StreamStats) {
avg_delay_ms_ = 1;
max_delay_ms_ = 2;
TEST_F(SendStatisticsProxyTest, SendSideDelay) {
SendSideDelayObserver* observer = statistics_proxy_.get();
for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
it != config_.rtp.ssrcs.end();
++it) {
const uint32_t ssrc = *it;
// Use ssrc as avg_delay_ms and max_delay_ms to get a unique value for each
// stream.
int avg_delay_ms = ssrc;
int max_delay_ms = ssrc + 1;
observer->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
expected_.substreams[ssrc].avg_delay_ms = avg_delay_ms;
expected_.substreams[ssrc].max_delay_ms = max_delay_ms;
}
VideoSendStream::Stats stats = statistics_proxy_->GetStats();
EXPECT_EQ(avg_delay_ms_, stats.avg_delay_ms);
EXPECT_EQ(max_delay_ms_, stats.max_delay_ms);
ExpectEqual(expected_, stats);
}
TEST_F(SendStatisticsProxyTest, NoSubstreams) {

31
webrtc/video/video_send_stream.cc
View File

@ -125,7 +125,7 @@ VideoSendStream::VideoSendStream(
suspended_ssrcs_(suspended_ssrcs),
external_codec_(NULL),
channel_(-1),
stats_proxy_(new SendStatisticsProxy(config, this)) {
stats_proxy_(config) {
video_engine_base_ = ViEBase::GetInterface(video_engine);
video_engine_base_->CreateChannel(channel_, base_channel);
assert(channel_ != -1);
@ -216,6 +216,8 @@ VideoSendStream::VideoSendStream(
if (overuse_observer)
video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
video_engine_base_->RegisterSendSideDelayObserver(channel_, &stats_proxy_);
image_process_ = ViEImageProcess::GetInterface(video_engine);
image_process_->RegisterPreEncodeCallback(channel_,
config_.pre_encode_callback);
@ -228,26 +230,26 @@ VideoSendStream::VideoSendStream(
codec_->SuspendBelowMinBitrate(channel_);
rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_,
stats_proxy_.get());
&stats_proxy_);
rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
stats_proxy_.get());
rtp_rtcp_->RegisterSendBitrateObserver(channel_, stats_proxy_.get());
rtp_rtcp_->RegisterSendFrameCountObserver(channel_, stats_proxy_.get());
&stats_proxy_);
rtp_rtcp_->RegisterSendBitrateObserver(channel_, &stats_proxy_);
rtp_rtcp_->RegisterSendFrameCountObserver(channel_, &stats_proxy_);
codec_->RegisterEncoderObserver(channel_, *stats_proxy_);
capture_->RegisterObserver(capture_id_, *stats_proxy_);
codec_->RegisterEncoderObserver(channel_, stats_proxy_);
capture_->RegisterObserver(capture_id_, stats_proxy_);
}
VideoSendStream::~VideoSendStream() {
capture_->DeregisterObserver(capture_id_);
codec_->DeregisterEncoderObserver(channel_);
rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, stats_proxy_.get());
rtp_rtcp_->DeregisterSendBitrateObserver(channel_, stats_proxy_.get());
rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, &stats_proxy_);
rtp_rtcp_->DeregisterSendBitrateObserver(channel_, &stats_proxy_);
rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_,
stats_proxy_.get());
&stats_proxy_);
rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_,
stats_proxy_.get());
&stats_proxy_);
image_process_->DeRegisterPreEncodeCallback(channel_);
@ -395,12 +397,7 @@ bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
}
VideoSendStream::Stats VideoSendStream::GetStats() const {
return stats_proxy_->GetStats();
}
bool VideoSendStream::GetSendSideDelay(VideoSendStream::Stats* stats) {
return codec_->GetSendSideDelay(
channel_, &stats->avg_delay_ms, &stats->max_delay_ms);
return stats_proxy_.GetStats();
}
void VideoSendStream::ConfigureSsrcs() {

9
webrtc/video/video_send_stream.h
View File

@ -38,8 +38,7 @@ class ViERTP_RTCP;
namespace internal {
class VideoSendStream : public webrtc::VideoSendStream,
public VideoSendStreamInput,
public SendStatisticsProxy::StatsProvider {
public VideoSendStreamInput {
public:
VideoSendStream(newapi::Transport* transport,
CpuOveruseObserver* overuse_observer,
@ -72,10 +71,6 @@ class VideoSendStream : public webrtc::VideoSendStream,
typedef std::map<uint32_t, RtpState> RtpStateMap;
RtpStateMap GetRtpStates() const;
protected:
// From SendStatisticsProxy::StreamStatsProvider.
virtual bool GetSendSideDelay(VideoSendStream::Stats* stats) OVERRIDE;
private:
void ConfigureSsrcs();
TransportAdapter transport_adapter_;
@ -96,7 +91,7 @@ class VideoSendStream : public webrtc::VideoSendStream,
int channel_;
int capture_id_;
const scoped_ptr<SendStatisticsProxy> stats_proxy_;
SendStatisticsProxy stats_proxy_;
};
} // namespace internal
} // namespace webrtc

8
webrtc/video/video_send_stream_tests.cc
View File

@ -27,6 +27,7 @@
#include "webrtc/system_wrappers/interface/scoped_vector.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/configurable_frame_size_encoder.h"
#include "webrtc/test/null_transport.h"
@ -956,8 +957,8 @@ TEST_F(VideoSendStreamTest, ProducesStats) {
bool CheckStats() {
VideoSendStream::Stats stats = stream_->GetStats();
// Check that all applicable data sources have been used.
if (stats.input_frame_rate > 0 && stats.encode_frame_rate > 0 &&
stats.avg_delay_ms > 0 && !stats.substreams.empty()) {
if (stats.input_frame_rate > 0 && stats.encode_frame_rate > 0
&& !stats.substreams.empty()) {
uint32_t ssrc = stats.substreams.begin()->first;
EXPECT_NE(
config_.rtp.ssrcs.end(),
@ -967,7 +968,8 @@ TEST_F(VideoSendStreamTest, ProducesStats) {
// data is received from remote side. Tested in call tests instead.
const StreamStats& entry = stats.substreams[ssrc];
if (entry.key_frames > 0u && entry.bitrate_bps > 0 &&
entry.rtp_stats.packets > 0u) {
entry.rtp_stats.packets > 0u && entry.avg_delay_ms > 0 &&
entry.max_delay_ms > 0) {
return true;
}
}

7
webrtc/video_engine/include/vie_base.h
View File

@ -204,6 +204,13 @@ class WEBRTC_DLLEXPORT ViEBase {
// Gets cpu overuse measures.
virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics) = 0;
// Registers a callback which is called when send-side delay statistics has
// been updated.
// TODO(holmer): Remove the default implementation when fakevideoengine.h has
// been updated.
virtual void RegisterSendSideDelayObserver(
int channel, SendSideDelayObserver* observer) {}
// Specifies the VoiceEngine and VideoEngine channel pair to use for
// audio/video synchronization.
virtual int ConnectAudioChannel(const int video_channel,

8
webrtc/video_engine/vie_base_impl.cc
View File

@ -144,6 +144,14 @@ int ViEBaseImpl::GetCpuOveruseMetrics(int video_channel,
return -1;
}
void ViEBaseImpl::RegisterSendSideDelayObserver(
int channel, SendSideDelayObserver* observer) {
ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
ViEChannel* vie_channel = cs.Channel(channel);
assert(vie_channel);
vie_channel->RegisterSendSideDelayObserver(observer);
}
int ViEBaseImpl::CreateChannel(int& video_channel) { // NOLINT
return CreateChannel(video_channel, static_cast<const Config*>(NULL));
}

2
webrtc/video_engine/vie_base_impl.h
View File

@ -37,6 +37,8 @@ class ViEBaseImpl
const CpuOveruseOptions& options);
virtual int GetCpuOveruseMetrics(int channel,
CpuOveruseMetrics* metrics);
virtual void RegisterSendSideDelayObserver(int channel,
SendSideDelayObserver* observer) OVERRIDE;
virtual int CreateChannel(int& video_channel); // NOLINT
virtual int CreateChannel(int& video_channel, // NOLINT
const Config* config);

7
webrtc/video_engine/vie_channel.cc
View File

@ -118,6 +118,7 @@ ViEChannel::ViEChannel(int32_t channel_id,
configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
configuration.send_bitrate_observer = &send_bitrate_observer_;
configuration.send_frame_count_observer = &send_frame_count_observer_;
configuration.send_side_delay_observer = &send_side_delay_observer_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
@ -1200,6 +1201,11 @@ bool ViEChannel::GetSendSideDelay(int* avg_send_delay,
return valid_estimate;
}
void ViEChannel::RegisterSendSideDelayObserver(
SendSideDelayObserver* observer) {
send_side_delay_observer_.Set(observer);
}
void ViEChannel::RegisterSendBitrateObserver(
BitrateStatisticsObserver* observer) {
send_bitrate_observer_.Set(observer);
@ -1558,6 +1564,7 @@ RtpRtcp* ViEChannel::CreateRtpRtcpModule() {
configuration.bandwidth_callback = bandwidth_observer_.get();
configuration.rtt_stats = rtt_stats_;
configuration.paced_sender = paced_sender_;
configuration.send_side_delay_observer = &send_side_delay_observer_;
return RtpRtcp::CreateRtpRtcp(configuration);
}

14
webrtc/video_engine/vie_channel.h
View File

@ -212,7 +212,10 @@ class ViEChannel
uint32_t* video_bitrate_sent,
uint32_t* fec_bitrate_sent,
uint32_t* nackBitrateSent) const;
// TODO(holmer): Deprecated. We should use the SendSideDelayObserver instead
// to avoid deadlocks.
bool GetSendSideDelay(int* avg_send_delay, int* max_send_delay) const;
void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
void GetReceiveBandwidthEstimatorStats(
ReceiveBandwidthEstimatorStats* output) const;
@ -429,6 +432,17 @@ class ViEChannel
}
} send_frame_count_observer_;
class RegisterableSendSideDelayObserver :
public RegisterableCallback<SendSideDelayObserver> {
virtual void SendSideDelayUpdated(int avg_delay_ms,
int max_delay_ms,
uint32_t ssrc) OVERRIDE {
CriticalSectionScoped cs(critsect_.get());
if (callback_)
callback_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
}
} send_side_delay_observer_;
int32_t channel_id_;
int32_t engine_id_;
uint32_t number_of_cores_;

5
webrtc/video_send_stream.h
View File

@ -41,14 +41,9 @@ class VideoSendStream {
Stats()
: input_frame_rate(0),
encode_frame_rate(0),
avg_delay_ms(0),
max_delay_ms(0),
suspended(false) {}
int input_frame_rate;
int encode_frame_rate;
int avg_delay_ms;
int max_delay_ms;
bool suspended;
std::map<uint32_t, StreamStats> substreams;
};