Add NACK rate throttling for audio channels.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
 #include <list>
 #include <map>
 #include <memory>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/system_wrappers/include/sleep.h"
@@ skipping 442 matching lines @@
     std::set<uint16_t> dropped_packets_;
     std::set<uint16_t> retransmitted_packets_;
     uint64_t sent_rtp_packets_;
     int packets_left_to_drop_;
     int nacks_left_ GUARDED_BY(&crit_);
   } test;
 
   RunBaseTest(&test);
 }
 
+TEST_F(EndToEndTest, ReceivesNackAndRetransmitsAudio) {
+  class NackObserver : public test::EndToEndTest {
+   public:
+    explicit NackObserver(
+        std::unique_ptr<test::PacketTransport>* send_transport)
+        : EndToEndTest(kLongTimeoutMs),
+          local_ssrc_(0),
+          remote_ssrc_(0),
+          send_transport_(send_transport) {}
+
+   private:
+    size_t GetNumVideoStreams() const override { return 0; }
+    size_t GetNumAudioStreams() const override { return 1; }
+
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+      if (!sequence_number_to_retransmit_) {
+        sequence_number_to_retransmit_ =
+            rtc::Optional<uint16_t>(header.sequenceNumber);
+
+        // Don't ask for retransmission straight away, may be deduped in pacer.
+      } else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
+        observation_complete_.Set();
+      } else {
+        // Send a NACK as often as necessary until retransmission is received.
+        rtcp::Nack nack;
+        nack.From(local_ssrc_);
+        nack.To(remote_ssrc_);
+        uint16_t nack_list[] = {*sequence_number_to_retransmit_};
+        nack.WithList(nack_list, 1);
+        rtc::Buffer buffer = nack.Build();
+
+        Transport* transport = static_cast<Transport*>(send_transport_->get());
+        EXPECT_TRUE(transport->SendRtcp(buffer.data(), buffer.size()));
+      }
+
+      return SEND_PACKET;
+    }
+
+    void ModifyAudioConfigs(
+        AudioSendStream::Config* send_config,
+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
+      remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait())
+          << "Timed out waiting for packets to be NACKed, retransmitted and "
+             "rendered.";
+    }
+
+    uint32_t local_ssrc_;
+    uint32_t remote_ssrc_;
+    std::unique_ptr<test::PacketTransport>* const send_transport_;

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

Doesn't it make more sense to store a test::PacketTransport* here? More clearly
shows that it's not owned by this class IMO.

[sprang_webrtc, 2016/07/28 15:17:18]

This is problematic, because this unique_ptr isn't populated when constructing
|test|; it's set up while running RunBaseTest()...

[stefan-webrtc, 2016/07/28 15:24:07]

Ah... In that case I'd suggest creating your own transport instead, see
EndToEndTest.RembWithSendSideBwe for an example of the receive transport
version:

    test::PacketTransport* CreateReceiveTransport() override {
      receive_transport_ = new test::PacketTransport(
          nullptr, this, test::PacketTransport::kReceiver,
          FakeNetworkPipe::Config());
      return receive_transport_;
    }

[sprang_webrtc, 2016/07/29 08:00:05]

Done.

+    rtc::Optional<uint16_t> sequence_number_to_retransmit_;
+  } test(&receive_transport_);
+
+  RunBaseTest(&test);
+}
+
 TEST_F(EndToEndTest, CanReceiveFec) {
   class FecRenderObserver : public test::EndToEndTest,
                             public rtc::VideoSinkInterface<VideoFrame> {
    public:
     FecRenderObserver()
         : EndToEndTest(kDefaultTimeoutMs), state_(kFirstPacket) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
@@ skipping 1300 matching lines @@
         : EndToEndTest(kDefaultTimeoutMs),
           sender_call_(nullptr),
           clock_(Clock::GetRealTimeClock()),
           sender_ssrc_(0),
           remb_bitrate_bps_(1000000),
           receive_transport_(nullptr),
           event_(false, false),
           poller_thread_(&BitrateStatsPollingThread,
                          this,
                          "BitrateStatsPollingThread"),
-          state_(kWaitForFirstRampUp) {}
+          state_(kWaitForFirstRampUp),
+          retransmission_rate_limiter_(clock_, 1000) {}

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

Why is this needed in this test now?

[sprang_webrtc, 2016/07/28 15:17:17]

Because I DCHECK in the RtpSender constructor that we always have a
retransmission rate limiter. Just so we don't miss any more cases...

[stefan-webrtc, 2016/07/28 15:24:07]

I see, that's probably a good idea.

 
     ~BweObserver() {}
 
     test::PacketTransport* CreateReceiveTransport() override {
       receive_transport_ = new test::PacketTransport(
           nullptr, this, test::PacketTransport::kReceiver,
           FakeNetworkPipe::Config());
       return receive_transport_;
     }
 
@@ skipping 19 matching lines @@
           encoder_config->streams[0].target_bitrate_bps = 2000000;
 
       ASSERT_EQ(1u, receive_configs->size());
       (*receive_configs)[0].rtp.remb = false;
       (*receive_configs)[0].rtp.transport_cc = true;
       (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
       RtpRtcp::Configuration config;
       config.receiver_only = true;
       config.clock = clock_;
       config.outgoing_transport = receive_transport_;
+      config.retransmission_rate_limiter = &retransmission_rate_limiter_;
       rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
       rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
       rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
       rtp_rtcp_->SetREMBStatus(true);
       rtp_rtcp_->SetSendingStatus(true);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
@@ skipping 52 matching lines @@
 
     Call* sender_call_;
     Clock* const clock_;
     uint32_t sender_ssrc_;
     int remb_bitrate_bps_;
     std::unique_ptr<RtpRtcp> rtp_rtcp_;
     test::PacketTransport* receive_transport_;
     rtc::Event event_;
     rtc::PlatformThread poller_thread_;
     TestState state_;
+    RateLimiter retransmission_rate_limiter_;
   } test;
 
   RunBaseTest(&test);
 }
 
 TEST_F(EndToEndTest, VerifyNackStats) {
   static const int kPacketNumberToDrop = 200;
   class NackObserver : public test::EndToEndTest {
    public:
     NackObserver()
@@ skipping 1713 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc