Adds logging of configuration information for VideoReceiveStream

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
deleted file mode 100644
index 98d0e622a871abe78d74678560d992993003f753..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- *  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