Added validation between RTP and RTCP timestamps

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 <map>
+#include <set>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
@@ skipping 2901 matching lines @@
     size_t ssrcs_to_observe_;
     std::map<uint32_t, bool> observed_redundant_retransmission_;
     std::map<uint32_t, bool> registered_rtx_ssrc_;
   } test;
 
   RunBaseTest(&test);
 }
 
 void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
   static const uint32_t kMaxSequenceNumberGap = 100;
-  static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
+  static const uint32_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_[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();
+      ssrc_observed_rtp_.clear();
+      ssrc_observed_rtcp_sr_.clear();
       ssrcs_to_observe_ = num_expected_ssrcs;
     }
 
    private:
+    void ValidateTimestampGap(uint32_t ssrc, uint32_t timestamp) {

[pbos-webrtc, 2016/01/28 10:25:06]

Can you use TimeDiff or TimeIsBetween in webrtc/base/timeutils.h to not
implement another roll-over?

[danilchap, 2016/01/28 10:44:09]

Thanks, that makes test more clear what is checked.

+      auto find_timestamp = last_observed_timestamp_.find(ssrc);
+      if (find_timestamp == last_observed_timestamp_.end()) {
+        last_observed_timestamp_[ssrc] = timestamp;
+      } else {
+        // Use unsigned operation to automagicaly manage roll-over.
+        uint32_t timestamp_gap = timestamp - find_timestamp->second;
+        if (timestamp_gap >= 0x80000000) {  // negative - negate.
+          timestamp_gap = (0xffffffff - timestamp_gap) + 1;
+        }
+        // It is normal for the gap to be negative: RTCP with current time
+        // can be sent just before RTP with capture time.
+        EXPECT_LE(timestamp_gap, kMaxTimestampGap)
+              << "Gap in timestamps ("
+              << find_timestamp->second
+              << " -> " << timestamp << ") too large for SSRC: " << ssrc << ".";
+        find_timestamp->second = timestamp;
+      }
+    }
+
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       const uint32_t ssrc = header.ssrc;
       const uint16_t sequence_number = header.sequenceNumber;
       const uint32_t timestamp = header.timestamp;
-      const bool only_padding =
-          header.headerLength + header.paddingLength == length;
 
       EXPECT_TRUE(configured_ssrcs_[ssrc])
           << "Received SSRC that wasn't configured: " << ssrc;
 
       std::map<uint32_t, uint16_t>::iterator it =
           last_observed_sequence_number_.find(header.ssrc);
       if (it == last_observed_sequence_number_.end()) {
         last_observed_sequence_number_[ssrc] = sequence_number;
-        last_observed_timestamp_[ssrc] = timestamp;
       } else {
         // Verify sequence numbers are reasonably close.
         uint32_t extended_sequence_number = sequence_number;
         // Check for roll-over.
         if (sequence_number < last_observed_sequence_number_[ssrc])
           extended_sequence_number += 0xFFFFu + 1;
         EXPECT_LE(
             extended_sequence_number - last_observed_sequence_number_[ssrc],
             kMaxSequenceNumberGap)
             << "Gap in sequence numbers ("
             << last_observed_sequence_number_[ssrc] << " -> " << sequence_number
             << ") too large for SSRC: " << ssrc << ".";
         last_observed_sequence_number_[ssrc] = sequence_number;
+      }
+      ValidateTimestampGap(ssrc, timestamp);
 
-        // TODO(pbos): Remove this check if we ever have monotonically
-        // increasing timestamps. Right now padding packets add a delta which
-        // can cause reordering between padding packets and regular packets,
-        // hence we drop padding-only packets to not flake.
-        if (only_padding) {
-          // Verify that timestamps are reasonably close.
-          uint64_t extended_timestamp = timestamp;
-          // Check for roll-over.
-          if (timestamp < last_observed_timestamp_[ssrc])
-            extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1;
-          EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc],
-                    kMaxTimestampGap)
-              << "Gap in timestamps (" << last_observed_timestamp_[ssrc]
-              << " -> " << timestamp << ") too large for SSRC: " << ssrc << ".";
-        }
-        last_observed_timestamp_[ssrc] = timestamp;
+      rtc::CritScope lock(&crit_);
+      ssrc_observed_rtp_.insert(ssrc);
+      if (ssrc_observed_rtcp_sr_.size() >= ssrcs_to_observe_ &&
+          ssrc_observed_rtp_.size() >= ssrcs_to_observe_) {
+        observation_complete_.Set();
       }
 
-      rtc::CritScope lock(&crit_);
-      // Wait for media packets on all ssrcs.
-      if (!ssrc_observed_[ssrc] && !only_padding) {
-        ssrc_observed_[ssrc] = true;
-        if (--ssrcs_to_observe_ == 0)
+      return SEND_PACKET;
+    }
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
+      test::RtcpPacketParser rtcp_parser;
+      rtcp_parser.Parse(packet, length);
+      if (rtcp_parser.sender_report()->num_packets() > 0) {
+        uint32_t ssrc = rtcp_parser.sender_report()->Ssrc();
+        uint32_t rtcp_timestamp = rtcp_parser.sender_report()->RtpTimestamp();
+        ValidateTimestampGap(ssrc, rtcp_timestamp);
+
+        rtc::CritScope lock(&crit_);
+        ssrc_observed_rtcp_sr_.insert(ssrc);
+        if (ssrc_observed_rtcp_sr_.size() >= ssrcs_to_observe_ &&
+            ssrc_observed_rtp_.size() >= ssrcs_to_observe_) {
           observation_complete_.Set();
+        }
       }
-
       return SEND_PACKET;
     }
 
     std::map<uint32_t, uint16_t> last_observed_sequence_number_;
     std::map<uint32_t, uint32_t> last_observed_timestamp_;
     std::map<uint32_t, bool> configured_ssrcs_;
 
     rtc::CriticalSection crit_;
     size_t ssrcs_to_observe_ GUARDED_BY(crit_);
-    std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
+    std::set<uint32_t> ssrc_observed_rtp_ GUARDED_BY(crit_);
+    std::set<uint32_t> ssrc_observed_rtcp_sr_ GUARDED_BY(crit_);
   } observer(use_rtx);
 
   CreateCalls(Call::Config(), Call::Config());
 
   test::PacketTransport send_transport(sender_call_.get(), &observer,
                                        test::PacketTransport::kSender,
                                        FakeNetworkPipe::Config());
   test::PacketTransport receive_transport(nullptr, &observer,
                                           test::PacketTransport::kReceiver,
                                           FakeNetworkPipe::Config());
@@ skipping 463 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc