Add NACK rate throttling for audio channels.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 424d291bcbb3a274dd5ce4a302bd014dad7bc5f4..db275e103d250335bc0a1cae2486a72ff4ba5a35 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -17,6 +17,7 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call/rtc_event_log.h"
@@ -47,6 +48,9 @@ namespace voe {
 
 namespace {
 
+constexpr int64_t kMaxRetranmissionWindowMs = 1000;
+constexpr int64_t kMinRetranmissionWindowMs = 30;
+
 bool RegisterReceiveCodec(std::unique_ptr<AudioCodingModule>* acm,
                           acm2::RentACodec* rac,
                           const CodecInst& ci) {
@@ -534,6 +538,13 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
   int64_t round_trip_time = 0;
   _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL,
                       NULL);
+  int64_t nack_window_ms = round_trip_time;
+  if (nack_window_ms < kMinRetranmissionWindowMs) {
+    nack_window_ms = kMinRetranmissionWindowMs;
+  } else if (nack_window_ms > kMaxRetranmissionWindowMs) {
+    nack_window_ms = kMaxRetranmissionWindowMs;
+  }
+  nack_rate_limiter_->SetWindowSize(nack_window_ms);

[stefan-webrtc, 2016/07/28 07:27:38]

This seems like a slightly odd place for this. Wouldn't it make more sense to do
this where an rtcp packet is recieved, as that actually may mean that the rtt is
updated?

[minyue-webrtc, 2016/07/28 08:17:27]

It might be good to put in VoERtcpObserver

[sprang_webrtc, 2016/07/28 13:00:58]

Done.

 
   std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time);
   if (!nack_list.empty()) {
@@ -902,6 +913,8 @@ Channel::Channel(int32_t channelId,
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
+      nack_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
+                                         kMaxRetranmissionWindowMs)),
       decoder_factory_(decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
@@ -933,6 +946,7 @@ Channel::Channel(int32_t channelId,
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
   configuration.event_log = &(*event_log_proxy_);
+  configuration.retransmission_rate_limiter = nack_rate_limiter_.get();

[stefan-webrtc, 2016/07/28 07:27:38]

should we rename it retransmission_rate_limiter_ instead of nack_rate_limiter_?

[sprang_webrtc, 2016/07/28 13:00:58]

Done.

 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
@@ -1352,6 +1366,7 @@ void Channel::SetBitRate(int bitrate_bps) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
   audio_coding_->SetBitRate(bitrate_bps);
+  nack_rate_limiter_->SetMaxRate(bitrate_bps);

[stefan-webrtc, 2016/07/28 07:27:38]

What will the default be if this is never called? Most of the time audio isn't
hooked up to the bitrate allocator.

[sprang_webrtc, 2016/07/28 13:00:58]

Rate limit defaults to maxint, so always allows retransimssion. That sound like
the right thing?

[stefan-webrtc, 2016/07/28 15:06:15]

Acknowledged.

 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {