Refactor NACK bitrate allocation

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/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_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"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
@@ skipping 1481 matching lines @@
 }
 
 class TransportFeedbackTester : public test::EndToEndTest {
  public:
   explicit TransportFeedbackTester(bool feedback_enabled,
                                    size_t num_video_streams,
                                    size_t num_audio_streams)
       : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
         feedback_enabled_(feedback_enabled),
         num_video_streams_(num_video_streams),
-        num_audio_streams_(num_audio_streams) {
+        num_audio_streams_(num_audio_streams),
+        receiver_call_(nullptr) {
     // Only one stream of each supported for now.
     EXPECT_LE(num_video_streams, 1u);
     EXPECT_LE(num_audio_streams, 1u);
   }
 
  protected:
   Action OnSendRtcp(const uint8_t* data, size_t length) override {
     EXPECT_FALSE(HasTransportFeedback(data, length));
     return SEND_PACKET;
   }
@@ skipping 974 matching lines @@
    public:
     StatsObserver()
         : EndToEndTest(kLongTimeoutMs),
           encoder_(Clock::GetRealTimeClock(), 10),
           send_stream_(nullptr),
           expected_send_ssrcs_(),
           check_stats_event_(false, false) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      const int kPacketLossFrac = 25;  // Drop every 25th packet => 4% loss.

[tommi, 2016/07/07 12:48:47]

should be packet_loss_frac or static const int kPacketLossFrac

[sprang_webrtc, 2016/07/07 13:46:51]

Done.

+      RTPHeader header;
+      RtpUtility::RtpHeaderParser parser(packet, length);
+      if (parser.Parse(&header) &&
+          expected_send_ssrcs_.find(header.ssrc) !=
+              expected_send_ssrcs_.end() &&
+          header.sequenceNumber % kPacketLossFrac == 0) {
+        return DROP_PACKET;
+      }
       check_stats_event_.Set();
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       check_stats_event_.Set();
       return SEND_PACKET;
     }
 
     Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
@@ skipping 80 matching lines @@
       send_stats_filled_["CpuOveruseMetrics"] |=
           stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0;
 
       send_stats_filled_["EncoderImplementationName"] |=
           stats.encoder_implementation_name ==
           test::FakeEncoder::kImplementationName;
 
       for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
                stats.substreams.begin();
            it != stats.substreams.end(); ++it) {
-        EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
-                    expected_send_ssrcs_.end());
+        if (expected_send_ssrcs_.find(it->first) == expected_send_ssrcs_.end())
+          continue;  // Probably RTX.
 
         send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
             stats.input_frame_rate != 0;
 
         const VideoSendStream::StreamStats& stream_stats = it->second;
 
         send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
             stream_stats.rtcp_stats.cumulative_lost != 0 ||
             stream_stats.rtcp_stats.extended_max_sequence_number != 0 ||
             stream_stats.rtcp_stats.fraction_lost != 0;
 
         send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
             stream_stats.rtp_stats.fec.packets != 0 ||
             stream_stats.rtp_stats.transmitted.padding_bytes != 0 ||
             stream_stats.rtp_stats.retransmitted.packets != 0 ||
             stream_stats.rtp_stats.transmitted.packets != 0;
 
-        send_stats_filled_[CompoundKey("BitrateStatisticsObserver",
+        send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Total",
                                        it->first)] |=
             stream_stats.total_bitrate_bps != 0;
 
+        send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Retransmit",
+                                       it->first)] |=
+            stream_stats.retransmit_bitrate_bps != 0;
+
         send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
             stream_stats.frame_counts.delta_frames != 0 ||
             stream_stats.frame_counts.key_frames != 0;
 
         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;
 
@@ skipping 10 matching lines @@
       return AllStatsFilled(send_stats_filled_);
     }
 
     std::string CompoundKey(const char* name, uint32_t ssrc) {
       std::ostringstream oss;
       oss << name << "_" << ssrc;
       return oss.str();
     }
 
     bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
-      for (std::map<std::string, bool>::const_iterator it = stats_map.begin();
-           it != stats_map.end();
-           ++it) {
-        if (!it->second)
+      for (const auto& stat : stats_map) {
+        if (!stat.second)
           return false;
       }
       return true;
     }
 
     test::PacketTransport* CreateSendTransport(Call* sender_call) override {
       FakeNetworkPipe::Config network_config;
       network_config.loss_percent = 5;
       return new test::PacketTransport(
           sender_call, this, test::PacketTransport::kSender, network_config);
     }
 
     Call::Config GetSenderCallConfig() override {
       Call::Config config = EndToEndTest::GetSenderCallConfig();
       config.bitrate_config.start_bitrate_bps = kStartBitrateBps;
       return config;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
+      // Set low rates to avoid waiting for rampup.
+      for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
+        encoder_config->streams[i].min_bitrate_bps = 10000;
+        encoder_config->streams[i].target_bitrate_bps = 15000;
+        encoder_config->streams[i].max_bitrate_bps = 20000;
+      }
       send_config->pre_encode_callback = this;  // Used to inject delay.
       expected_cname_ = send_config->rtp.c_name = "SomeCName";
 
+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
+
       const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
       for (size_t i = 0; i < ssrcs.size(); ++i) {
         expected_send_ssrcs_.insert(ssrcs[i]);
         expected_receive_ssrcs_.push_back(
             (*receive_configs)[i].rtp.remote_ssrc);
         (*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
         (*receive_configs)[i].renderer = &receive_stream_renderer_;
+        (*receive_configs)[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+
+        (*receive_configs)[i].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
+            kSendRtxSsrcs[i];
+        (*receive_configs)[i].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
+            kSendRtxPayloadType;
       }
+
+      for (size_t i = 0; i < kNumSsrcs; ++i)
+        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
+
       // Use a delayed encoder to make sure we see CpuOveruseMetrics stats that
       // are non-zero.
       send_config->encoder_settings.encoder = &encoder_;
     }
 
     size_t GetNumVideoStreams() const override { return kNumSsrcs; }
 
     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
@@ skipping 833 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc