Enable probing with repeated payload packets by default.

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 364 matching lines @@
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
 
       // Never drop retransmitted packets.
       if (dropped_packets_.find(header.sequenceNumber) !=
           dropped_packets_.end()) {
         retransmitted_packets_.insert(header.sequenceNumber);
-        if (nacks_left_ == 0 &&
+        if (nacks_left_ <= 0 &&
             retransmitted_packets_.size() == dropped_packets_.size()) {
           observation_complete_->Set();
         }
         return SEND_PACKET;
       }
 
       ++sent_rtp_packets_;
 
       // Enough NACKs received, stop dropping packets.
-      if (nacks_left_ == 0)
+      if (nacks_left_ <= 0)
         return SEND_PACKET;
 
       // Check if it's time for a new loss burst.
       if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
         packets_left_to_drop_ = kLossBurstSize;
 
       // Never drop padding packets as those won't be retransmitted.
       if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
         --packets_left_to_drop_;
         dropped_packets_.insert(header.sequenceNumber);
@@ skipping 265 matching lines @@
     bool has_last_sequence_number_;
     uint16_t last_sequence_number_;
   } test(config);
 
   RunBaseTest(&test);
 }
 
 // This test drops second RTP packet with a marker bit set, makes sure it's
 // retransmitted and renders. Retransmission SSRCs are also checked.
 void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
-  static const int kDroppedFrameNumber = 2;
+  // 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_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 {
       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(payload_type_, header.payloadType);
 
-      // Found the second frame's final packet, drop this and expect a
-      // retransmission.
+      // 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;
     }
 
     void FrameCallback(VideoFrame* frame) override {
       rtc::CritScope lock(&crit_);
@@ skipping 34 matching lines @@
 
     int GetPayloadType(bool use_rtx, bool use_red) {
       return use_rtx ? kSendRtxPayloadType
                      : (use_red ? kRedPayloadType : kFakeSendPayloadType);
     }
 
     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_;
     bool frame_retransmitted_;
   } test(use_rtx, use_red);
 
   RunBaseTest(&test);
 }
 
 TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
   DecodesRetransmittedFrame(false, false);
 }
@@ skipping 2272 matching lines @@
   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);
 }
 
 }  // namespace webrtc