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Wire up bandwidth stats to the new API and webrtcvideoengine2.

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Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2014-11-05 14:05:29 +00:00
parent a22a628356
commit 0bae1fab4a
26 changed files with 315 additions and 129 deletions
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80
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -40,6 +40,57 @@ const char* FrameTypeToString(const FrameType frame_type) {
} // namespace
class BitrateAggregator {
public:
explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback)
: callback_(bitrate_callback),
total_bitrate_observer_(*this),
retransmit_bitrate_observer_(*this),
ssrc_(0) {}
void OnStatsUpdated() const {
if (callback_)
callback_->Notify(total_bitrate_observer_.statistics(),
retransmit_bitrate_observer_.statistics(),
ssrc_);
}
Bitrate::Observer* total_bitrate_observer() {
return &total_bitrate_observer_;
}
Bitrate::Observer* retransmit_bitrate_observer() {
return &retransmit_bitrate_observer_;
}
void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
private:
// We assume that these observers are called on the same thread, which is
// true for RtpSender as they are called on the Process thread.
class BitrateObserver : public Bitrate::Observer {
public:
explicit BitrateObserver(const BitrateAggregator& aggregator)
: aggregator_(aggregator) {}
// Implements Bitrate::Observer.
virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE {
statistics_ = stats;
aggregator_.OnStatsUpdated();
}
BitrateStatistics statistics() const { return statistics_; }
private:
BitrateStatistics statistics_;
const BitrateAggregator& aggregator_;
};
BitrateStatisticsObserver* const callback_;
BitrateObserver total_bitrate_observer_;
BitrateObserver retransmit_bitrate_observer_;
uint32_t ssrc_;
};
RTPSender::RTPSender(const int32_t id,
const bool audio,
Clock* clock,
@ -54,7 +105,8 @@ RTPSender::RTPSender(const int32_t id,
// TickTime.
clock_delta_ms_(clock_->TimeInMilliseconds() -
TickTime::MillisecondTimestamp()),
bitrate_sent_(clock, this),
bitrates_(new BitrateAggregator(bitrate_callback)),
total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
id_(id),
audio_configured_(audio),
audio_(NULL),
@ -74,12 +126,11 @@ RTPSender::RTPSender(const int32_t id,
// NACK.
nack_byte_count_times_(),
nack_byte_count_(),
nack_bitrate_(clock, NULL),
nack_bitrate_(clock, bitrates_->retransmit_bitrate_observer()),
packet_history_(clock),
// Statistics
statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
rtp_stats_callback_(NULL),
bitrate_callback_(bitrate_callback),
frame_count_observer_(frame_count_observer),
send_side_delay_observer_(send_side_delay_observer),
// RTP variables
@ -108,6 +159,7 @@ RTPSender::RTPSender(const int32_t id,
srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
@ -149,7 +201,7 @@ uint32_t RTPSender::GetTargetBitrate() {
}
uint16_t RTPSender::ActualSendBitrateKbit() const {
return (uint16_t)(bitrate_sent_.BitrateNow() / 1000);
return (uint16_t)(total_bitrate_sent_.BitrateNow() / 1000);
}
uint32_t RTPSender::VideoBitrateSent() const {
@ -864,7 +916,7 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer,
counters = &rtp_stats_;
}
bitrate_sent_.Update(size);
total_bitrate_sent_.Update(size);
++counters->packets;
if (IsFecPacket(buffer, header)) {
++counters->fec_packets;
@ -997,7 +1049,7 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
void RTPSender::ProcessBitrate() {
CriticalSectionScoped cs(send_critsect_);
bitrate_sent_.Process();
total_bitrate_sent_.Process();
nack_bitrate_.Process();
if (audio_configured_) {
return;
@ -1420,6 +1472,7 @@ void RTPSender::SetSendingStatus(bool enabled) {
// Generate a new SSRC.
ssrc_db_.ReturnSSRC(ssrc_);
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
}
// Don't initialize seq number if SSRC passed externally.
if (!sequence_number_forced_ && !ssrc_forced_) {
@ -1470,6 +1523,7 @@ uint32_t RTPSender::GenerateNewSSRC() {
return 0;
}
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
return ssrc_;
}
@ -1484,6 +1538,7 @@ void RTPSender::SetSSRC(uint32_t ssrc) {
ssrc_db_.ReturnSSRC(ssrc_);
ssrc_db_.RegisterSSRC(ssrc);
ssrc_ = ssrc;
bitrates_->set_ssrc(ssrc_);
if (!sequence_number_forced_) {
sequence_number_ =
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
@ -1681,17 +1736,8 @@ StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
return rtp_stats_callback_;
}
uint32_t RTPSender::BitrateSent() const { return bitrate_sent_.BitrateLast(); }
void RTPSender::BitrateUpdated(const BitrateStatistics& stats) {
uint32_t ssrc;
{
CriticalSectionScoped ssrc_lock(send_critsect_);
ssrc = ssrc_;
}
if (bitrate_callback_) {
bitrate_callback_->Notify(stats, ssrc);
}
uint32_t RTPSender::BitrateSent() const {
return total_bitrate_sent_.BitrateLast();
}
void RTPSender::SetRtpState(const RtpState& rtp_state) {