Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific t…

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
new file mode 100644
index 0000000000000000000000000000000000000000..98d0e622a871abe78d74678560d992993003f753
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
@@ -0,0 +1,124 @@
+/*
+ *  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.
+ */
+
+#ifdef RTC_AUDIOCODING_DEBUG_DUMP
+
+#include <stdio.h>
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_dump.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/test/test_suite.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h"
+#else
+#include "webrtc/audio_coding/dump.pb.h"
+#endif
+
+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:
+  void VerifyResults(const ACMDumpEventStream& parsed_stream,
+                     size_t packet_size) {
+    // Verify the result.
+    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 = 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) {
+        EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction());
+      } 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();
+
+    // When log_dumper goes out of scope, it causes the log file to be flushed
+    // to disk.
+    {
+      rtc::scoped_ptr<AcmDump> log_dumper(AcmDump::Create());
+      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.
+    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_;
+};
+
+TEST_F(AcmDumpTest, DumpAndRead) {
+  Run(256, 321);
+}
+
+}  // namespace webrtc
+
+#endif  // RTC_AUDIOCODING_DEBUG_DUMP