Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.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.
 */
 
@@ skipping 91 matching lines @@
     count_++;
     const unsigned char* ptr = static_cast<const unsigned  char*>(data);
     uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
     if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
     uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
     size_t packet_length = len;
     // TODO(pbos): Figure out why this needs to be initialized. Likely this
     // is hiding a bug either in test setup or other code.
     // https://code.google.com/p/webrtc/issues/detail?id=3183
     uint8_t restored_packet[1500] = {0};
-    uint8_t* restored_packet_ptr = restored_packet;
     RTPHeader header;
     rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     if (!parser->Parse(ptr, len, &header)) {
       return false;
     }
 
     if (!rtp_payload_registry_->IsRtx(header)) {
       // Don't store retransmitted packets since we compare it to the list
       // created by the receiver.
       expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
                                         sequence_number);
     }
     if (packet_loss_ > 0) {
       if ((count_ % packet_loss_) == 0) {
         return true;
       }
     } else if (count_ >= consecutive_drop_start_ &&
                count_ < consecutive_drop_end_) {
       return true;
     }
     if (rtp_payload_registry_->IsRtx(header)) {
       // Remove the RTX header and parse the original RTP header.
       EXPECT_TRUE(rtp_payload_registry_->RestoreOriginalPacket(
-          &restored_packet_ptr, ptr, &packet_length, rtp_receiver_->SSRC(),
-          header));
-      if (!parser->Parse(restored_packet_ptr, packet_length, &header)) {
+          restored_packet, ptr, &packet_length, rtp_receiver_->SSRC(), header));
+      if (!parser->Parse(restored_packet, packet_length, &header)) {
         return false;
       }
     } else {
       rtp_payload_registry_->SetIncomingPayloadType(header);
     }
 
-    restored_packet_ptr += header.headerLength;
+    const uint8_t* restored_packet_payload =
+        restored_packet + header.headerLength;
     packet_length -= header.headerLength;
     PayloadUnion payload_specific;
     if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                     &payload_specific)) {
       return false;
     }
-    if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_ptr,
+    if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_payload,
                                           packet_length, payload_specific,
                                           true)) {
       return false;
     }
     return true;
   }
 
   bool SendRtcp(const uint8_t* data, size_t len) override {
     return module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0;
   }
@@ skipping 181 matching lines @@
 
 TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
       *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }