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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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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);
}
};