Wire-up BWE feedback for audio receive streams.

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video/call_stats.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
 
 AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
   AudioDecodingCallStats audio_decode_stats;
   audio_decode_stats.calls_to_silence_generator = 234;
   audio_decode_stats.calls_to_neteq = 567;
   audio_decode_stats.decoded_normal = 890;
   audio_decode_stats.decoded_plc = 123;
   audio_decode_stats.decoded_cng = 456;
   audio_decode_stats.decoded_plc_cng = 789;
   return audio_decode_stats;
 }
 
 const int kChannelId = 2;
 const uint32_t kRemoteSsrc = 1234;
 const uint32_t kLocalSsrc = 5678;
-const size_t kAbsoluteSendTimeLength = 4;
+const size_t kOneByteExtensionLength = 4;
 const int kAbsSendTimeId = 2;
 const int kAudioLevelId = 3;
+const int kTransportSequenceNumberId = 4;
 const int kJitterBufferDelay = -7;
 const int kPlayoutBufferDelay = 302;
 const unsigned int kSpeechOutputLevel = 99;
 const CallStatistics kCallStats = {
     345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
 const CodecInst kCodecInst = {
     123, "codec_name_recv", 96000, -187, -198, -103};
 const NetworkStatistics kNetworkStats = {
     123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
 
 struct ConfigHelper {
-  ConfigHelper() {
+  ConfigHelper()
+      : call_stats_(Clock::GetRealTimeClock()),
+        process_thread_(ProcessThread::Create("AudioTestThread")),

[the sun, 2015/12/22 00:14:14]

Are we relying on this thread to run somehow? Or is it just to satisfy the CC
c-tor? Can we use a mock instead?

[stefan-webrtc, 2016/01/07 13:43:41]

Done.

+        congestion_controller_(process_thread_.get(),
+                               &call_stats_,
+                               &bitrate_observer_) {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
+    channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
 
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
-          EXPECT_FALSE(channel_proxy_);
-          channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
                   .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
   }
 
+  MockCongestionController* congestion_controller() {
+    return &congestion_controller_;
+  }
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
 
+  void SetupMockForBweFeedback() {
+    EXPECT_CALL(congestion_controller_, GetRemoteBitrateEstimator(true))
+        .WillOnce(Return(&remote_bitrate_estimator_));
+    EXPECT_CALL(congestion_controller_, packet_router())
+        .WillOnce(Return(&packet_router_));
+    EXPECT_CALL(remote_bitrate_estimator_,
+                RemoveStream(stream_config_.rtp.remote_ssrc));
+    ASSERT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_,
+                SetCongestionControlObjects(nullptr, nullptr, &packet_router_))
+        .Times(1);
+    EXPECT_CALL(*channel_proxy_,
+                SetCongestionControlObjects(nullptr, nullptr, nullptr))
+        .Times(1);
+  }
+
   void SetupMockForGetStats() {
     using testing::DoAll;
     using testing::SetArgReferee;
 
-    EXPECT_TRUE(channel_proxy_);
+    ASSERT_TRUE(channel_proxy_);

[the sun, 2015/12/22 00:14:14]

thanks

     EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
         .WillOnce(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
         .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
     EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
         .WillOnce(Return(kSpeechOutputLevel));
     EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
         .WillOnce(Return(kNetworkStats));
     EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
         .WillOnce(Return(kAudioDecodeStats));
 
     EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
         .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
   }
 
  private:
+  CallStats call_stats_;
+  PacketRouter packet_router_;
+  testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+  rtc::scoped_ptr<ProcessThread> process_thread_;
+  MockCongestionController congestion_controller_;
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioReceiveStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };
 
-void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
-                                    int id,
-                                    uint32_t abs_send_time) {
+void BuildOneByteExtension(uint8_t* buffer,

[the sun, 2015/12/22 00:14:14]

vector<uint8_t>::iterator?

+                           int id,
+                           uint32_t extension_value,
+                           size_t value_length) {
   const size_t kRtpOneByteHeaderLength = 4;

[the sun, 2015/12/22 00:14:14]

Can this be removed?

[stefan-webrtc, 2016/01/07 13:43:41]

Done.

   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
   ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
 
   const uint32_t kPosLength = 2;
   ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
-                                       kAbsoluteSendTimeLength / 4);
-
-  const uint8_t kLengthOfData = 3;
-  buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
-  ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
-      buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
+                                       kOneByteExtensionLength / 4);
+  const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
+  uint32_t shifted_value = extension_value
+                           << (8 * (kExtensionDataLength - value_length));
+  buffer[kRtpOneByteHeaderLength] = (id << 4) + (value_length - 1);
+  ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(
+      buffer + kRtpOneByteHeaderLength + 1, shifted_value);
 }
 
-size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
-                                      int extension_id,
-                                      uint32_t abs_send_time) {
+size_t CreateRtpHeaderWithOneByteExtension(uint8_t* header,

[the sun, 2015/12/22 00:14:14]

How about just returning a vector<uint8_t> by value. Safer.

[stefan-webrtc, 2016/01/07 13:43:41]

Done.

+                                           int extension_id,
+                                           uint32_t extension_value,
+                                           size_t value_length) {
   header[0] = 0x80;   // Version 2.
   header[0] |= 0x10;  // Set extension bit.
   header[1] = 100;    // Payload type.
   header[1] |= 0x80;  // Marker bit is set.
   ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234);  // Sequence number.
   ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678);  // Timestamp.
   ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321);  // SSRC.
   int32_t rtp_header_length = webrtc::kRtpHeaderSize;
 
-  BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
-                                 abs_send_time);
-  rtp_header_length += kAbsoluteSendTimeLength;
+  BuildOneByteExtension(header + rtp_header_length, extension_id,
+                        extension_value, value_length);
+  rtp_header_length += kOneByteExtensionLength;
   return rtp_header_length;
 }
 }  // namespace
 
 TEST(AudioReceiveStreamTest, ConfigToString) {
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
   config.rtp.local_ssrc = kLocalSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   config.voe_channel_id = kChannelId;
   config.combined_audio_video_bwe = true;
   EXPECT_EQ(
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
       "receive_transport: nullptr, rtcp_send_transport: nullptr, "
       "voe_channel_id: 2, combined_audio_video_bwe: true}",
       config.ToString());
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
-      helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+}
+
+MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
+  return arg.extension.hasAbsoluteSendTime ==
+             expected_extension.hasAbsoluteSendTime &&
+         arg.extension.absoluteSendTime ==
+             expected_extension.absoluteSendTime &&
+         arg.extension.hasTransportSequenceNumber ==
+             expected_extension.hasTransportSequenceNumber &&
+         arg.extension.transportSequenceNumber ==
+             expected_extension.transportSequenceNumber;
 }
 
 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
   ConfigHelper helper;
   helper.config().combined_audio_video_bwe = true;
+  EXPECT_CALL(*helper.congestion_controller(), GetRemoteBitrateEstimator(false))
+      .WillOnce(Return(helper.remote_bitrate_estimator()));
+  EXPECT_CALL(*helper.remote_bitrate_estimator(),
+              RemoveStream(helper.config().rtp.remote_ssrc));
   internal::AudioReceiveStream recv_stream(
-      helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state());
   uint8_t rtp_packet[30];
   const int kAbsSendTimeValue = 1234;
-  CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
+  CreateRtpHeaderWithOneByteExtension(rtp_packet, kAbsSendTimeId,
+                                      kAbsSendTimeValue, 3);
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
+  RTPHeaderExtension expected_extension;
+  expected_extension.hasAbsoluteSendTime = true;
+  expected_extension.absoluteSendTime = kAbsSendTimeValue;
   EXPECT_CALL(*helper.remote_bitrate_estimator(),
               IncomingPacket(packet_time.timestamp / 1000,
                              sizeof(rtp_packet) - kExpectedHeaderLength,
-                             testing::_, false))
+                             VerifyHeaderExtension(expected_extension), false))
       .Times(1);
   EXPECT_TRUE(
       recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
+}
+
+TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
+  ConfigHelper helper;
+  helper.config().combined_audio_video_bwe = true;
+  helper.config().rtp.transport_cc = true;
+  helper.config().rtp.extensions.push_back(RtpExtension(
+      RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+  helper.SetupMockForBweFeedback();
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+  uint8_t rtp_packet[30];
+  const int kTransportSequenceNumberValue = 1234;
+  CreateRtpHeaderWithOneByteExtension(rtp_packet, kTransportSequenceNumberId,
+                                      kTransportSequenceNumberValue, 2);
+  PacketTime packet_time(5678000, 0);
+  const size_t kExpectedHeaderLength = 20;
+  RTPHeaderExtension expected_extension;
+  expected_extension.hasTransportSequenceNumber = true;
+  expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
+  EXPECT_CALL(*helper.remote_bitrate_estimator(),
+              IncomingPacket(packet_time.timestamp / 1000,
+                             sizeof(rtp_packet) - kExpectedHeaderLength,
+                             VerifyHeaderExtension(expected_extension), false))
+      .Times(1);
+  EXPECT_TRUE(
+      recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
 }
 
 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
-      helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state());
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
   EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
             stats.packets_rcvd);
   EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
   EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
   EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
@@ skipping 19 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc