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

webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc

Index: webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
index 55c948ebf72905dba4dadd5dd4ede9b6ab5961d8..b9ba99fbf196329d4cdb514fc039010d84591c08 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
@@ -39,8 +39,8 @@ class AcmDumpTest : public ::testing::Test {
   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());
@@ -51,7 +51,11 @@ class AcmDumpTest : public ::testing::Test {
     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());
@@ -60,9 +64,9 @@ class AcmDumpTest : public ::testing::Test {
       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());
@@ -88,8 +92,10 @@ class AcmDumpTest : public ::testing::Test {
     const std::string temp_filename =
         test::OutputPath() + test_info->test_case_name() + test_info->name();
 
-    log_dumper_->StartLogging(temp_filename, 10000000);
     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(false, rtp_packet_.data(), rtp_packet_.size());
     log_dumper_->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
 
     // Read the generated file from disk.
@@ -109,7 +115,6 @@ class AcmDumpTest : public ::testing::Test {
 
 TEST_F(AcmDumpTest, DumpAndRead) {
   Run(256, 321);
-  Run(256, 123);
 }
 
 }  // namespace webrtc