Add audio streams to CallTest and a first A/V call test.

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>
@@ skipping 13 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.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/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.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"
-#include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/histogram.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
@@ skipping 35 matching lines @@
   void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
   void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
   void TestRtpStatePreservation(bool use_rtx);
   void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
 };
 
 TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
   CreateCalls(Call::Config(), Call::Config());
 
   test::NullTransport transport;
-  CreateSendConfig(1, &transport);
-  CreateMatchingReceiveConfigs(&transport);
+  CreateSendConfig(1, 0, &transport);
+  CreateMatchingReceiveConfigs(false, &transport);
 
-  CreateStreams();
+  CreateVideoStreams();
 
   video_receive_streams_[0]->Start();
   video_receive_streams_[0]->Start();
 
   DestroyStreams();
 }
 
 TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) {
   CreateCalls(Call::Config(), Call::Config());
 
   test::NullTransport transport;
-  CreateSendConfig(1, &transport);
-  CreateMatchingReceiveConfigs(&transport);
+  CreateSendConfig(1, 0, &transport);
+  CreateMatchingReceiveConfigs(false, &transport);
 
-  CreateStreams();
+  CreateVideoStreams();
 
   video_receive_streams_[0]->Stop();
   video_receive_streams_[0]->Stop();
 
   DestroyStreams();
 }
 
 TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
   static const int kWidth = 320;
   static const int kHeight = 240;
@@ skipping 33 matching lines @@
     rtc::Event event_;
   };
 
   CreateCalls(Call::Config(), Call::Config());
 
   test::DirectTransport sender_transport(sender_call_.get());
   test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1, &sender_transport);
-  CreateMatchingReceiveConfigs(&receiver_transport);
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(false, &receiver_transport);
 
   TestFrameCallback pre_render_callback;
   video_receive_configs_[0].pre_render_callback = &pre_render_callback;
   video_receive_configs_[0].renderer = &renderer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(kWidth, kHeight));
   video_send_stream_->Input()->IncomingCapturedFrame(
       *frame_generator->NextFrame());
   EXPECT_TRUE(pre_render_callback.Wait())
       << "Timed out while waiting for pre-render callback.";
@@ skipping 24 matching lines @@
     rtc::Event event_;
   } renderer;
 
   CreateCalls(Call::Config(), Call::Config());
 
   test::DirectTransport sender_transport(sender_call_.get());
   test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1, &sender_transport);
-  CreateMatchingReceiveConfigs(&receiver_transport);
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(false, &receiver_transport);
   video_receive_configs_[0].renderer = &renderer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(
           video_encoder_config_.streams[0].width,
           video_encoder_config_.streams[0].height));
   video_send_stream_->Input()->IncomingCapturedFrame(
       *frame_generator->NextFrame());
 
   EXPECT_TRUE(renderer.Wait())
@@ skipping 73 matching lines @@
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->encoder_settings.encoder = &fake_encoder_;
       send_config->encoder_settings.payload_name = "H264";
-      send_config->encoder_settings.payload_type = kFakeSendPayloadType;
+      send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
           encoder_config->streams[0].max_bitrate_bps = 2000000;
 
       (*receive_configs)[0].renderer = this;
       (*receive_configs)[0].decoders.resize(1);
       (*receive_configs)[0].decoders[0].payload_type =
           send_config->encoder_settings.payload_type;
       (*receive_configs)[0].decoders[0].payload_name =
           send_config->encoder_settings.payload_name;
@@ skipping 24 matching lines @@
     SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
       uint32_t ssrc = 0;
       ssrc |= static_cast<uint32_t>(packet[4]) << 24;
       ssrc |= static_cast<uint32_t>(packet[5]) << 16;
       ssrc |= static_cast<uint32_t>(packet[6]) << 8;
       ssrc |= static_cast<uint32_t>(packet[7]) << 0;
-      EXPECT_EQ(kReceiverLocalSsrc, ssrc);
+      EXPECT_EQ(kReceiverLocalVideoSsrc, ssrc);
       observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for a receiver RTCP packet to be sent.";
     }
   } test;
@@ skipping 100 matching lines @@
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       int encapsulated_payload_type = -1;
       if (header.payloadType == kRedPayloadType) {
         encapsulated_payload_type =
             static_cast<int>(packet[header.headerLength]);
-        if (encapsulated_payload_type != kFakeSendPayloadType)
+        if (encapsulated_payload_type != kFakeVideoSendPayloadType)
           EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
       } else {
-        EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
+        EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
       }
 
       if (protected_sequence_numbers_.count(header.sequenceNumber) != 0) {
         // Retransmitted packet, should not count.
         protected_sequence_numbers_.erase(header.sequenceNumber);
         EXPECT_GT(protected_timestamps_.count(header.timestamp), 0u);
         protected_timestamps_.erase(header.timestamp);
         return SEND_PACKET;
       }
 
       switch (state_) {
         case kFirstPacket:
           state_ = kDropEveryOtherPacketUntilFec;
           break;
         case kDropEveryOtherPacketUntilFec:
           if (encapsulated_payload_type == kUlpfecPayloadType) {
             state_ = kDropNextMediaPacket;
             return SEND_PACKET;
           }
           if (header.sequenceNumber % 2 == 0)
             return DROP_PACKET;
           break;
         case kDropNextMediaPacket:
-          if (encapsulated_payload_type == kFakeSendPayloadType) {
+          if (encapsulated_payload_type == kFakeVideoSendPayloadType) {
             protected_sequence_numbers_.insert(header.sequenceNumber);
             protected_timestamps_.insert(header.timestamp);
             state_ = kDropEveryOtherPacketUntilFec;
             return DROP_PACKET;
           }
           break;
       }
 
       return SEND_PACKET;
     }
@@ skipping 58 matching lines @@
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock_(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       int encapsulated_payload_type = -1;
       if (header.payloadType == kRedPayloadType) {
         encapsulated_payload_type =
             static_cast<int>(packet[header.headerLength]);
-        if (encapsulated_payload_type != kFakeSendPayloadType)
+        if (encapsulated_payload_type != kFakeVideoSendPayloadType)
           EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
       } else {
-        EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
+        EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
       }
 
       if (has_last_sequence_number_ &&
           !IsNewerSequenceNumber(header.sequenceNumber,
                                  last_sequence_number_)) {
         // Drop retransmitted packets.
         return DROP_PACKET;
       }
       last_sequence_number_ = header.sequenceNumber;
       has_last_sequence_number_ = true;
@@ skipping 94 matching lines @@
 void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
   // Must be set high enough to allow the bitrate probing to finish.
   static const int kMinProbePackets = 30;
   static const int kDroppedFrameNumber = kMinProbePackets + 1;
   class RetransmissionObserver : public test::EndToEndTest,
                                  public I420FrameCallback {
    public:
     explicit RetransmissionObserver(bool use_rtx, bool use_red)
         : EndToEndTest(kDefaultTimeoutMs),
           payload_type_(GetPayloadType(false, use_red)),
-          retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
+          retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kVideoSendSsrcs[0]),
           retransmission_payload_type_(GetPayloadType(use_rtx, use_red)),
           marker_bits_observed_(0),
           num_packets_observed_(0),
           retransmitted_timestamp_(0),
           frame_retransmitted_(false) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       // We accept some padding or RTX packets in the beginning to enable
       // bitrate probing.
       if (num_packets_observed_++ < kMinProbePackets &&
           header.payloadType != payload_type_) {
         EXPECT_TRUE(retransmission_payload_type_ == header.payloadType ||
                     length == header.headerLength + header.paddingLength);
         return SEND_PACKET;
       }
       if (header.timestamp == retransmitted_timestamp_) {
         EXPECT_EQ(retransmission_ssrc_, header.ssrc);
         EXPECT_EQ(retransmission_payload_type_, header.payloadType);
         frame_retransmitted_ = true;
         return SEND_PACKET;
       }
 
-      EXPECT_EQ(kSendSsrcs[0], header.ssrc);
+      EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc);
       EXPECT_EQ(payload_type_, header.payloadType);
 
       // Found the final packet of the frame to inflict loss to, drop this and
       // expect a retransmission.
       if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) {
         retransmitted_timestamp_ = header.timestamp;
         return DROP_PACKET;
       }
 
       return SEND_PACKET;
@@ skipping 18 matching lines @@
       if (payload_type_ == kRedPayloadType) {
         send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
         send_config->rtp.fec.red_payload_type = kRedPayloadType;
         (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
         (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
       }
 
       if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
-        (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
             kSendRtxSsrcs[0];
-        (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
             kSendRtxPayloadType;
       }
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for retransmission to render.";
     }
 
     int GetPayloadType(bool use_rtx, bool use_red) {
       return use_rtx ? kSendRtxPayloadType
-                     : (use_red ? kRedPayloadType : kFakeSendPayloadType);
+                     : (use_red ? kRedPayloadType : kFakeVideoSendPayloadType);
     }
 
     rtc::CriticalSection crit_;
     const int payload_type_;
     const uint32_t retransmission_ssrc_;
     const int retransmission_payload_type_;
     int marker_bits_observed_;
     int num_packets_observed_;
     uint32_t retransmitted_timestamp_ GUARDED_BY(&crit_);
     bool frame_retransmitted_;
@@ skipping 76 matching lines @@
   TestFrameCallback pre_encode_callback(-1, 255);  // Changes luma to 255.
   TestFrameCallback pre_render_callback(255, 0);  // Changes luma from 255 to 0.
 
   CreateCalls(Call::Config(), Call::Config());
 
   test::DirectTransport sender_transport(sender_call_.get());
   test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1, &sender_transport);
+  CreateSendConfig(1, 0, &sender_transport);
   rtc::scoped_ptr<VideoEncoder> encoder(
       VideoEncoder::Create(VideoEncoder::kVp8));
   video_send_config_.encoder_settings.encoder = encoder.get();
   video_send_config_.encoder_settings.payload_name = "VP8";
   ASSERT_EQ(1u, video_encoder_config_.streams.size()) << "Test setup error.";
   video_encoder_config_.streams[0].width = kWidth;
   video_encoder_config_.streams[0].height = kHeight;
   video_send_config_.pre_encode_callback = &pre_encode_callback;
 
-  CreateMatchingReceiveConfigs(&receiver_transport);
+  CreateMatchingReceiveConfigs(false, &receiver_transport);
   video_receive_configs_[0].pre_render_callback = &pre_render_callback;
   video_receive_configs_[0].renderer = &renderer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(kWidth / 2, kHeight / 2));
   video_send_stream_->Input()->IncomingCapturedFrame(
       *frame_generator->NextFrame());
 
   EXPECT_TRUE(pre_encode_callback.Wait())
@@ skipping 139 matching lines @@
   };
 
   CreateCalls(Call::Config(), Call::Config());
 
   test::DirectTransport send_transport(sender_call_.get());
   test::DirectTransport receive_transport(receiver_call_.get());
   PacketInputObserver input_observer(receiver_call_->Receiver());
   send_transport.SetReceiver(&input_observer);
   receive_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1, &send_transport);
-  CreateMatchingReceiveConfigs(&receive_transport);
+  CreateSendConfig(1, 0, &send_transport);
+  CreateMatchingReceiveConfigs(false, &receive_transport);
 
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
   Start();
 
   receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
   video_receive_streams_.clear();
 
   // Wait() waits for a received packet.
   EXPECT_TRUE(input_observer.Wait());
 
   Stop();
@@ skipping 153 matching lines @@
           stream->max_bitrate_bps = 100000;
 
       UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
 
       send_streams[i] =
           sender_call->CreateVideoSendStream(send_config, encoder_config);
       send_streams[i]->Start();
 
       VideoReceiveStream::Config receive_config(receiver_transport.get());
       receive_config.rtp.remote_ssrc = ssrc;
-      receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalSsrc;
+      receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
       VideoReceiveStream::Decoder decoder =
           test::CreateMatchingDecoder(send_config.encoder_settings);
       allocated_decoders.push_back(decoder.decoder);
       receive_config.decoders.push_back(decoder);
 
       UpdateReceiveConfig(i, &receive_config);
 
       receive_streams[i] =
           receiver_call->CreateVideoReceiveStream(receive_config);
       receive_streams[i]->Start();
@@ skipping 411 matching lines @@
   EncodedFrameTestObserver post_encode_observer;
   EncodedFrameTestObserver pre_decode_observer;
 
   CreateCalls(Call::Config(), Call::Config());
 
   test::DirectTransport sender_transport(sender_call_.get());
   test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1, &sender_transport);
-  CreateMatchingReceiveConfigs(&receiver_transport);
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(false, &receiver_transport);
   video_send_config_.post_encode_callback = &post_encode_observer;
   video_receive_configs_[0].pre_decode_callback = &pre_decode_observer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(
           video_encoder_config_.streams[0].width,
           video_encoder_config_.streams[0].height));
   video_send_stream_->Input()->IncomingCapturedFrame(
       *frame_generator->NextFrame());
 
   EXPECT_TRUE(post_encode_observer.Wait())
@@ skipping 20 matching lines @@
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       bool received_psfb = false;
       bool received_remb = false;
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
       while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
         if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRemb) {
           const RTCPUtility::RTCPPacket& packet = parser.Packet();
-          EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
+          EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalVideoSsrc);
           received_psfb = true;
         } else if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRembItem) {
           const RTCPUtility::RTCPPacket& packet = parser.Packet();
           EXPECT_GT(packet.REMBItem.BitRate, 0u);
           EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
-          EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]);
+          EXPECT_EQ(packet.REMBItem.SSRCs[0], kVideoSendSsrcs[0]);
           received_remb = true;
         }
         packet_type = parser.Iterate();
       }
       if (received_psfb && received_remb)
         observation_complete_.Set();
       return SEND_PACKET;
     }
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
@@ skipping 93 matching lines @@
            stats.substreams.begin(); it != stats.substreams.end(); ++it) {
         const VideoSendStream::StreamStats& stream_stats = it->second;
         send_stream_nack_packets +=
             stream_stats.rtcp_packet_type_counts.nack_packets;
       }
       for (size_t i = 0; i < receive_streams_.size(); ++i) {
         VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
         receive_stream_nack_packets +=
             stats.rtcp_packet_type_counts.nack_packets;
       }
-      if (send_stream_nack_packets >= 1 &&
-          receive_stream_nack_packets >= 1) {
+      if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
         // NACK packet sent on receive stream and received on sent stream.
         if (MinMetricRunTimePassed())
           observation_complete_.Set();
       }
     }
 
     bool MinMetricRunTimePassed() {
       int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
       if (start_runtime_ms_ == -1) {
         start_runtime_ms_ = now;
@@ skipping 93 matching lines @@
       if (use_red_) {
         send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
         send_config->rtp.fec.red_payload_type = kRedPayloadType;
         (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
         (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
       }
       // RTX
       if (use_rtx_) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
-        (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
             kSendRtxSsrcs[0];
-        (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
             kSendRtxPayloadType;
       }
       encoder_config->content_type =
           screenshare_ ? VideoEncoderConfig::ContentType::kScreen
                        : VideoEncoderConfig::ContentType::kRealtimeVideo;
     }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
@@ skipping 273 matching lines @@
           observation_complete_.Set();
         }
       }
 
       if (ssrcs_to_observe_ == 0)
         observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
-    size_t GetNumStreams() const override { return num_ssrcs_; }
+    size_t GetNumVideoStreams() const override { return num_ssrcs_; }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       if (num_ssrcs_ > 1) {
         // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
         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;
@@ skipping 30 matching lines @@
     std::map<uint32_t, bool> is_observed_;
 
     const size_t num_ssrcs_;
     const bool send_single_ssrc_first_;
 
     size_t ssrcs_to_observe_;
     bool expect_single_ssrc_;
 
     VideoSendStream* send_stream_;
     VideoEncoderConfig video_encoder_config_all_streams_;
-  } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);
+  } test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 TEST_F(EndToEndTest, ReportsSetEncoderRates) {
   class EncoderRateStatsTest : public test::EndToEndTest,
                                public test::FakeEncoder {
    public:
     EncoderRateStatsTest()
         : EndToEndTest(kDefaultTimeoutMs),
@@ skipping 135 matching lines @@
         receive_stats_filled_["CName"] |= !stats.c_name.empty();
 
         receive_stats_filled_["RtcpPacketTypeCount"] |=
             stats.rtcp_packet_type_counts.fir_packets != 0 ||
             stats.rtcp_packet_type_counts.nack_packets != 0 ||
             stats.rtcp_packet_type_counts.pli_packets != 0 ||
             stats.rtcp_packet_type_counts.nack_requests != 0 ||
             stats.rtcp_packet_type_counts.unique_nack_requests != 0;
 
         assert(stats.current_payload_type == -1 ||
-               stats.current_payload_type == kFakeSendPayloadType);
+               stats.current_payload_type == kFakeVideoSendPayloadType);
         receive_stats_filled_["IncomingPayloadType"] |=
-            stats.current_payload_type == kFakeSendPayloadType;
+            stats.current_payload_type == kFakeVideoSendPayloadType;
       }
 
       return AllStatsFilled(receive_stats_filled_);
     }
 
     bool CheckSendStats() {
       RTC_DCHECK(send_stream_ != nullptr);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       send_stats_filled_["NumStreams"] |=
@@ skipping 86 matching lines @@
 
       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;
       }
     }
 
-    size_t GetNumStreams() const override { return kNumSsrcs; }
+    size_t GetNumVideoStreams() const override { return kNumSsrcs; }
 
     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
       receive_streams_ = receive_streams;
     }
 
     void PerformTest() override {
       Clock* clock = Clock::GetRealTimeClock();
@@ skipping 140 matching lines @@
 
       if (!observed_redundant_retransmission_[header.ssrc]) {
         observed_redundant_retransmission_[header.ssrc] = true;
         if (--ssrcs_to_observe_ == 0)
           observation_complete_.Set();
       }
 
       return SEND_PACKET;
     }
 
-    size_t GetNumStreams() const override { return kNumSsrcs; }
+    size_t GetNumVideoStreams() const override { return kNumSsrcs; }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
       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;
@@ skipping 25 matching lines @@
 
 void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
   static const uint32_t kMaxSequenceNumberGap = 100;
   static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
   class RtpSequenceObserver : public test::RtpRtcpObserver {
    public:
     explicit RtpSequenceObserver(bool use_rtx)
         : test::RtpRtcpObserver(kDefaultTimeoutMs),
           ssrcs_to_observe_(kNumSsrcs) {
       for (size_t i = 0; i < kNumSsrcs; ++i) {
-        configured_ssrcs_[kSendSsrcs[i]] = true;
+        configured_ssrcs_[kVideoSendSsrcs[i]] = true;
         if (use_rtx)
           configured_ssrcs_[kSendRtxSsrcs[i]] = true;
       }
     }
 
     void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
       rtc::CritScope lock(&crit_);
       ssrc_observed_.clear();
       ssrcs_to_observe_ = num_expected_ssrcs;
     }
@@ skipping 72 matching lines @@
 
   test::PacketTransport send_transport(sender_call_.get(), &observer,
                                        test::PacketTransport::kSender,
                                        FakeNetworkPipe::Config());
   test::PacketTransport receive_transport(nullptr, &observer,
                                           test::PacketTransport::kReceiver,
                                           FakeNetworkPipe::Config());
   send_transport.SetReceiver(receiver_call_->Receiver());
   receive_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(kNumSsrcs, &send_transport);
+  CreateSendConfig(kNumSsrcs, 0, &send_transport);
 
   if (use_rtx) {
     for (size_t i = 0; i < kNumSsrcs; ++i) {
       video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
     }
     video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
   }
 
   // Lower bitrates so that all streams send initially.
   for (size_t i = 0; i < video_encoder_config_.streams.size(); ++i) {
     video_encoder_config_.streams[i].min_bitrate_bps = 10000;
     video_encoder_config_.streams[i].target_bitrate_bps = 15000;
     video_encoder_config_.streams[i].max_bitrate_bps = 20000;
   }
 
   // Use the same total bitrates when sending a single stream to avoid lowering
   // the bitrate estimate and requiring a subsequent rampup.
   VideoEncoderConfig one_stream = video_encoder_config_;
   one_stream.streams.resize(1);
   for (size_t i = 1; i < video_encoder_config_.streams.size(); ++i) {
     one_stream.streams.front().min_bitrate_bps +=
         video_encoder_config_.streams[i].min_bitrate_bps;
     one_stream.streams.front().target_bitrate_bps +=
         video_encoder_config_.streams[i].target_bitrate_bps;
     one_stream.streams.front().max_bitrate_bps +=
         video_encoder_config_.streams[i].max_bitrate_bps;
   }
 
-  CreateMatchingReceiveConfigs(&receive_transport);
+  CreateMatchingReceiveConfigs(false, &receive_transport);
 
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
   EXPECT_TRUE(observer.Wait())
       << "Timed out waiting for all SSRCs to send packets.";
 
   // Test stream resetting more than once to make sure that the state doesn't
   // get set once (this could be due to using std::map::insert for instance).
   for (size_t i = 0; i < 3; ++i) {
     frame_generator_capturer_->Stop();
@@ skipping 225 matching lines @@
   CreateCalls(Call::Config(), Call::Config());
 
   FakeNetworkPipe::Config config;
   config.queue_delay_ms = kSendDelayMs;
   test::DirectTransport sender_transport(config, sender_call_.get());
   config.queue_delay_ms = kReceiveDelayMs;
   test::DirectTransport receiver_transport(config, receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1, &sender_transport);
-  CreateMatchingReceiveConfigs(&receiver_transport);
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(false, &receiver_transport);
 
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
   Start();
 
   int64_t start_time_ms = clock_->TimeInMilliseconds();
   while (true) {
     Call::Stats stats = sender_call_->GetStats();
     ASSERT_GE(start_time_ms + kDefaultTimeoutMs,
               clock_->TimeInMilliseconds())
         << "No RTT stats before timeout!";
     if (stats.rtt_ms != -1) {
@@ skipping 17 matching lines @@
       ADD_FAILURE() << "Unexpected frame encode.";
       return test::FakeEncoder::Encode(
           input_image, codec_specific_info, frame_types);
     }
   };
 
   CreateSenderCall(Call::Config());
   sender_call_->SignalNetworkState(kNetworkDown);
 
   UnusedTransport transport;
-  CreateSendConfig(1, &transport);
+  CreateSendConfig(1, 0, &transport);
   UnusedEncoder unused_encoder;
   video_send_config_.encoder_settings.encoder = &unused_encoder;
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
   SleepMs(kSilenceTimeoutMs);
   Stop();
 
   DestroyStreams();
 }
 
 TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
   CreateCalls(Call::Config(), Call::Config());
   receiver_call_->SignalNetworkState(kNetworkDown);
 
   test::DirectTransport sender_transport(sender_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
-  CreateSendConfig(1, &sender_transport);
+  CreateSendConfig(1, 0, &sender_transport);
   UnusedTransport transport;
-  CreateMatchingReceiveConfigs(&transport);
-  CreateStreams();
+  CreateMatchingReceiveConfigs(false, &transport);
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
   SleepMs(kSilenceTimeoutMs);
   Stop();
 
   sender_transport.StopSending();
 
   DestroyStreams();
 }
@@ skipping 36 matching lines @@
       << "RTCP XR settings require rtcp-xr to be negotiated.";
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
+#if defined(WEBRTC_ANDROID)

[pbos-webrtc, 2016/01/07 14:23:37]

Re-enable on Android, look into the isolate files to have this run.

[stefan-webrtc, 2016/01/07 15:16:44]

Done.

+#define MAYBE_TransportSeqNumOnAudioAndVideo DISABLED_TransportSeqNumOnAudioAndVideo
+#else
+#define MAYBE_TransportSeqNumOnAudioAndVideo TransportSeqNumOnAudioAndVideo
+#endif
+TEST_F(EndToEndTest, MAYBE_TransportSeqNumOnAudioAndVideo) {
+  static const int kExtensionId = 8;
+  class TransportSequenceNumberTest : public test::EndToEndTest {
+   public:
+    TransportSequenceNumberTest()
+        : EndToEndTest(kDefaultTimeoutMs),
+          video_observed_(false),
+          audio_observed_(false) {
+      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                          kExtensionId);
+    }
+
+    size_t GetNumVideoStreams() const override { return 1; }
+    size_t GetNumAudioStreams() const override { return 1; }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();

[pbos-webrtc, 2016/01/07 14:23:37]

(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; here and
below

[stefan-webrtc, 2016/01/07 15:16:44]

Done.

+      (*receive_configs)[0].rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    void ModifyAudioConfigs(
+        AudioSendStream::Config* send_config,
+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();
+      (*receive_configs)[0].rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+      // Unwrap packet id and verify uniqueness.
+      int64_t packet_id =
+          unwrapper_.Unwrap(header.extension.transportSequenceNumber);
+      EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
+
+      if (header.ssrc == kVideoSendSsrcs[0])
+        video_observed_ = true;
+      if (header.ssrc == kAudioSendSsrc)
+        audio_observed_ = true;
+      if (audio_observed_ && video_observed_ &&
+          received_packet_ids_.size() == 50) {
+        size_t packet_id_range =
+            *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
+        EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
+        observation_complete_.Set();
+      }
+      return SEND_PACKET;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
+                             "packets with transport sequence number.";
+    }
+
+   private:
+    bool video_observed_;
+    bool audio_observed_;
+    SequenceNumberUnwrapper unwrapper_;
+    std::set<int64_t> received_packet_ids_;
+  } test;
+
+  RunBaseTest(&test, FakeNetworkPipe::Config());
+}
 }  // namespace webrtc