Added support for keeping a buffer of the previous X seconds, to add to

webrtc/modules/audio_coding/main/acm2/acm_dump_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.
  */
 
@@ skipping 21 matching lines @@
 namespace webrtc {
 
 // Test for the acm dump class. Dumps some RTP packets to disk, then reads them
 // back to see if they match.
 class AcmDumpTest : public ::testing::Test {
  public:
   AcmDumpTest() : log_dumper_(AcmDump::Create()) {}
   void VerifyResults(const ACMDumpEventStream& parsed_stream,
                      size_t packet_size) {
     // Verify the result.
-    EXPECT_EQ(3, parsed_stream.stream_size());
-    const ACMDumpEvent& start_event = parsed_stream.stream(0);
+    EXPECT_EQ(5, parsed_stream.stream_size());
+    const ACMDumpEvent& start_event = parsed_stream.stream(2);
     ASSERT_TRUE(start_event.has_type());
     EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, start_event.type());
     EXPECT_TRUE(start_event.has_timestamp_us());
     EXPECT_FALSE(start_event.has_packet());
     ASSERT_TRUE(start_event.has_debug_event());
     auto start_debug_event = start_event.debug_event();
     ASSERT_TRUE(start_debug_event.has_type());
     EXPECT_EQ(ACMDumpDebugEvent::LOG_START, start_debug_event.type());
     ASSERT_TRUE(start_debug_event.has_message());
 
-    for (int i = 1; i < parsed_stream.stream_size(); i++) {
+    for (int i = 0; i < parsed_stream.stream_size(); i++) {
+      if (i==2) {
+        // This is the LOG_START packet that was already verified.
+        continue;
+      }
       const ACMDumpEvent& test_event = parsed_stream.stream(i);
       ASSERT_TRUE(test_event.has_type());
       EXPECT_EQ(ACMDumpEvent::RTP_EVENT, test_event.type());
       EXPECT_TRUE(test_event.has_timestamp_us());
       EXPECT_FALSE(test_event.has_debug_event());
       ASSERT_TRUE(test_event.has_packet());
       const ACMDumpRTPPacket& test_packet = test_event.packet();
       ASSERT_TRUE(test_packet.has_direction());
-      if (i == 1) {
+      if (i <= 1) {
         EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction());
-      } else if (i == 2) {
+      } else if (i >= 3) {
         EXPECT_EQ(ACMDumpRTPPacket::OUTGOING, test_packet.direction());
       }
       ASSERT_TRUE(test_packet.has_rtp_data());
       ASSERT_EQ(packet_size, test_packet.rtp_data().size());
       for (size_t i = 0; i < packet_size; i++) {
         EXPECT_EQ(rtp_packet_[i],
                   static_cast<uint8_t>(test_packet.rtp_data()[i]));
       }
     }
   }
 
   void Run(int packet_size, int random_seed) {
     rtp_packet_.clear();
     rtp_packet_.reserve(packet_size);
     srand(random_seed);
     // Fill the packet vector with random data.
     for (int i = 0; i < packet_size; i++) {
       rtp_packet_.push_back(rand());
     }
     // Find the name of the current test, in order to use it as a temporary
     // filename.
     auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
     const std::string temp_filename =
         test::OutputPath() + test_info->test_case_name() + test_info->name();
 
+    log_dumper_->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
+    log_dumper_->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
     log_dumper_->StartLogging(temp_filename, 10000000);
-    log_dumper_->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
+    log_dumper_->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
     log_dumper_->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
 
     // Read the generated file from disk.
     ACMDumpEventStream parsed_stream;
 
     ASSERT_EQ(true, AcmDump::ParseAcmDump(temp_filename, &parsed_stream));
 
     VerifyResults(parsed_stream, packet_size);
 
     // Clean up temporary file - can be pretty slow.
     remove(temp_filename.c_str());
   }
 
   std::vector<uint8_t> rtp_packet_;
   rtc::scoped_ptr<AcmDump> log_dumper_;
 };
 
 TEST_F(AcmDumpTest, DumpAndRead) {
   Run(256, 321);
-  Run(256, 123);
 }
 
 }  // namespace webrtc
 
 #endif  // RTC_AUDIOCODING_DEBUG_DUMP